U.S. patent application number 15/122835 was filed with the patent office on 2017-05-18 for method and apparatus for improving call quality of hands-free call device, and hands-free call device.
The applicant listed for this patent is QINGDAO GOERTEK TECHNOLOGY CO., LTD.. Invention is credited to Yang Hua, Hongwei Zhou.
Application Number | 20170142243 15/122835 |
Document ID | / |
Family ID | 51505197 |
Filed Date | 2017-05-18 |
United States Patent
Application |
20170142243 |
Kind Code |
A1 |
Hua; Yang ; et al. |
May 18, 2017 |
METHOD AND APPARATUS FOR IMPROVING CALL QUALITY OF HANDS-FREE CALL
DEVICE, AND HANDS-FREE CALL DEVICE
Abstract
The present invention discloses a method and an apparatus for
improving the call quality of a hands-free call device, and a
hands-free call device. The hands-free call device comprises a
transmitter end composed of a main microphone and at least one
auxiliary microphone. The method comprises: scanning within an
initial first collection angle of a transmitter end; after a voice
feature signal is scanned within the first collection angle,
according to a direction of the voice feature signal, determining a
second collection angle smaller than the first collection angle
within the first collection angle; and calibrating the transmitter
end to a direction determined by the second collection angle. In
this method, voice pickup is conducted using a relatively small
voice protection angle, so that the interference from ambient noise
can be greatly reduced, to achieve the purpose of improving the
transmission signal to noise ratio, thus the directivity of a voice
in a hands-free call device during a call is more clear, which
improves the call quality.
Inventors: |
Hua; Yang; (Qingdao City,
CN) ; Zhou; Hongwei; (Qingdao City, CN) |
|
Applicant: |
Name |
City |
State |
Country |
Type |
QINGDAO GOERTEK TECHNOLOGY CO., LTD. |
Qingado City |
|
CN |
|
|
Family ID: |
51505197 |
Appl. No.: |
15/122835 |
Filed: |
June 29, 2015 |
PCT Filed: |
June 29, 2015 |
PCT NO: |
PCT/CN2015/082634 |
371 Date: |
August 31, 2016 |
Current U.S.
Class: |
1/1 |
Current CPC
Class: |
H04M 1/6033 20130101;
G10L 2021/02166 20130101; H04M 1/03 20130101; H04R 3/005
20130101 |
International
Class: |
H04M 1/60 20060101
H04M001/60; H04M 1/03 20060101 H04M001/03 |
Foreign Application Data
Date |
Code |
Application Number |
Jun 30, 2014 |
CN |
201410307430.8 |
Claims
1. A method for improving the call quality of a hands-free call
device that comprises a transmitter end composed of a main
microphone and at least one auxiliary microphone, wherein the
method comprises: scanning a voice feature signal within an initial
first collection angle of the transmitter end; after a voice
feature signal is scanned within the first collection angle,
according to a direction of the voice feature signal, determining a
second collection angle smaller than the first collection angle
within the first collection angle; and calibrating the transmitter
end to a direction determined by the second collection angle;
wherein after a voice feature signal is scanned within the first
collection angle, according to a direction of the voice feature
signal, determining a second collection angle smaller than the
first collection angle within the first collection angle comprises:
making a reverse extension line through the main microphone in
direction .gamma.1 when a voice feature signal is scanned in
direction .gamma.1 within the initial first collection angle of the
transmitter end, drawing a circle by taking the main microphone as
a center and a connection line between the main microphone and one
of the auxiliary microphones as a radius, and determining
intersection of an arc of the circle and the reverse extension line
as a virtual microphone of the main microphone; taking the main
microphone and its virtual microphone as a new voice array,
defining an angle .beta.1 smaller than the first collection angle,
and judging in real time whether a voice feature signal is existed
within the angle .beta.1; if so, determining the angle .beta.1 as a
second collection angle; and if not, taking the connection line
between the main microphone and one of the auxiliary microphones as
an axis of symmetry, defining a mirror angle .beta.2 of the angle
.beta.1 with respect to the axis of symmetry, and determining the
angle .beta.2 as a second collection angle.
2. (canceled)
3. The method according to claim 1, wherein judging in real time
whether a voice feature signal is existed within the angle .beta.1
comprises: detecting envelope energy of the voice feature signal in
direction .gamma.1, and detecting a zero-crossing rate of the voice
feature signal in direction .gamma.1 when an energy detection value
is larger than a first predetermined threshold; and determining
that a voice feature signal is existed in the angle .beta.1, when
the zero-crossing rate of the voice feature signal in direction
.gamma.1 reaches a second predetermined threshold.
4. The method according to claim 3, wherein detecting envelope
energy of the voice feature signal in direction .gamma.1 comprises:
detecting envelope energy through the following formula:
power=0;power=power*(1-alpha)+.SIGMA..sub.n=1.sup.Nx(n)*x(n)
wherein, power is an energy value of the voice feature signal,
parameter alpha is a weighted factor, and parameter N is a specific
value of voice feature signal at a time point; detecting a
zero-crossing rate of the voice feature signal in direction
.gamma.1 comprises: detecting a zero-crossing rate through the
following formula: Z_rate = 1 2 m = 0 N - 1 | sgn [ x ( n ) - sgn (
x ( n - 1 ) ) ] ##EQU00009## wherein, Z_rate is a zero-crossing
rate of the voice feature signal, n is a value among a discrete
time series, and sgn [ x ] = { 1 , x > 0 - 1 , x < 0 } .
##EQU00010##
5. The method according to claim 1, wherein the hands-free call
device further comprises a receiver end composed of at least one
loudspeaker, and wherein the method further comprises: providing a
virtual loudspeaker of one loudspeaker in the receiver end with a
connection line between the loudspeaker and its virtual loudspeaker
directed to direction .gamma.1, and defining a third collection
angle having a regional extent covering a direction determined by
the second collection angle.
6. An apparatus for improving the call quality of a hands-free call
device that comprises a transmitter end composed of a main
microphone and at least one auxiliary microphone, comprising: a
voice feature determination unit configured to scan a voice feature
signal within an initial first collection angle of a transmitter
end; and after a voice feature signal is scanned within the first
collection angle, according to a direction of the voice feature
signal, determine a second collection angle smaller than the first
collection angle within the first collection angle; and a direction
calibration unit configured to calibrate the transmitter end to a
direction determined by the second collection angle; wherein the
voice feature determination unit comprising: a virtual microphone
creation unit configured to make a reverse extension line through
the main microphone in direction .gamma.1 when a voice feature
signal is scanned in direction .gamma.1 within the initial first
collection angle of the transmitter end, draw a circle by taking
the main microphone as a center and a connection line between the
main microphone and one of the auxiliary microphones as a radius,
and determine intersection of an arc of the circle and the reverse
extension line as a virtual microphone of the main microphone; and
an angle determination unit configured to take the main microphone
and its virtual microphone as a new voice array, define an angle
.beta.1 smaller than the first collection angle, and judge in real
time whether a voice feature signal is existed within the angle
.beta.1; if so, determine the angle .beta.1 as a second collection
angle; and if not, take the connection line between the main
microphone and one of the auxiliary microphones as an axis of
symmetry, define a mirror angle .beta.2 of the angle .beta.1 with
respect to the axis of symmetry, and determine the angle .beta.2 as
a second collection angle.
7. (canceled)
8. The apparatus according to claim 6, wherein the angle
determination unit is further configured to: detect envelope energy
of the voice feature signal in direction .gamma.1, and detect a
zero-crossing rate of the voice feature signal in direction
.gamma.1 when an energy detection value is larger than a first
predetermined threshold; and determine that a voice feature signal
is existed in the angle .beta.1, when the zero-crossing rate of the
voice feature signal in direction .gamma.1 reaches a second
predetermined threshold; the angle determination unit comprises: an
envelope detection unit configured to detect envelope energy of the
voice feature signal in direction .gamma.1 through the following
formula:
power=0;power=power*(1-alpha)+.SIGMA..sub.n=1.sup.Nx(n)*x(n)
wherein, power is an energy value of the voice feature signal,
parameter alpha is a weighted factor, and parameter N is a specific
value of a voice feature signal at a time point; and a
zero-crossing detection unit configured to detect a zero-crossing
rate of the voice feature signal in direction .gamma.1 through the
following formula: Z_rate = 1 2 m = 0 N - 1 | sgn [ x ( n ) - sgn (
x ( n - 1 ) ) ] ##EQU00011## wherein, Z_rate is a zero-crossing
rate of the voice feature signal, n is a value among a discrete
time series, and sgn [ x ] = { 1 , x > 0 - 1 , x < 0 } .
##EQU00012##
9. The apparatus according to claim 6, wherein the hands-free call
device further comprises a receiver end composed of at least one
loudspeaker, and wherein the apparatus further comprising: a
receiver end location unit configured to provide a virtual
loudspeaker of one loudspeaker in the receiver end with a
connection line between the loudspeaker and its virtual loudspeaker
directed to direction .gamma.1, and define a third collection angle
having a regional extent covering a direction determined by the
second collection angle.
10. A hands-free call device, comprising a transmitter end composed
of a main microphone and at least one auxiliary microphone, a
receiver end composed of at least one loudspeaker, and the
apparatus according to claim 6.
Description
FIELD OF THE INVENTION
[0001] The present invention relates to the field of hands-free
call devices, and particularly, to a method and an apparatus for
improving the call quality of a hands-free call device, and a
hands-free call device.
BACKGROUND OF THE INVENTION
[0002] Hands-free call devices in the prior art, such as smart
wrist-wearing devices like smart watch and smart band, usually use
a rather large voice protection angle under the hands-free call
mode since the position of the smart watch relative to a user's
mouth is uncertain, thus much ambient noise is collected during the
voice pickup, and the transmission signal to noise ratio is
influenced. Meanwhile, sound from the loudspeaker of the hands-free
call device such as the smart watch will be heard by not only the
caller himself, but also other people nearby, and the private
information can be easily leaked. In conclusion, the above defects
of the transmitter end and the receiver end lead to not so high a
call quality of the hands-free call device in the prior art.
SUMMARY OF THE INVENTION
[0003] The present invention provides a method and an apparatus for
improving the call quality of a hands-free call device, and a
hands-free call device, so as to solve the problem that much
ambient noise is collected and the transmission signal to noise
ratio is low during a call of the hands-free call device.
[0004] According to one aspect of the present invention, a method
for improving the call quality of a hands-free call device is
provided, wherein the hands-free call device comprises a
transmitter end composed of a main microphone and at least one
auxiliary microphone, and the method comprises:
[0005] scanning within an initial first collection angle of the
transmitter end;
[0006] after a voice feature signal is scanned within the first
collection angle, according to a direction of the voice feature
signal, determining a second collection angle smaller than the
first collection angle within the first collection angle; and
[0007] calibrating the transmitter end to a direction determined by
the second collection angle.
[0008] Wherein, after a voice feature signal is scanned within the
first collection angle, according to a direction of the voice
feature signal, determining a second collection angle smaller than
the first collection angle within the first collection angle
comprises:
[0009] making a reverse extension line through the main microphone
in direction .gamma.1 when a voice feature signal is scanned in
direction .gamma.1 within the initial first collection angle of the
transmitter end, drawing a circle by taking the main microphone as
a center and a connection line between the main microphone and one
of the auxiliary microphones as a radius, and determining
intersection of an arc of the circle and the reverse extension line
as a virtual microphone of the main microphone;
[0010] taking the main microphone and its virtual microphone as a
new voice array, defining an angle .beta.1 smaller than the first
collection angle, and judging in real time whether a voice feature
signal is existed within the angle .beta.1; if so, determining the
angle .beta.1 as a second collection angle; and
[0011] if not, taking the connection line between the main
microphone and one of the auxiliary microphones as an axis of
symmetry, defining a mirror angle .beta.2 of the angle .beta.1 with
respect to the axis of symmetry, and determining the angle .beta.2
as a second collection angle.
[0012] Wherein, judging in real time whether a voice feature signal
is existed within the angle .beta.1 comprises:
[0013] detecting envelope energy of the voice feature signal in
direction .gamma.1, and detecting a zero-crossing rate of the voice
feature signal in direction .gamma.1 when an energy detection value
is larger than a first predetermined threshold; and
[0014] determining that a voice feature signal is existed in the
angle .beta.1, when the zero-crossing rate of the voice feature
signal in direction .gamma.1 reaches a second predetermined
threshold.
[0015] Wherein, detecting envelope energy of the voice feature
signal in direction .gamma.1 comprises:
[0016] detecting envelope energy through the following formula:
power=0;power=power*(1-alpha)+.SIGMA..sub.n=1.sup.Nx(n)*x(n)
[0017] wherein, power is an energy value of the voice feature
signal, parameter alpha is a weighted factor, and parameter N is a
specific value of voice feature signal at a time point;
[0018] detecting a zero-crossing rate of the voice feature signal
in direction .gamma.1 comprises: detecting a zero-crossing rate
through the following formula:
Z_rate = 1 2 m = 0 N - 1 | sgn [ x ( n ) - sgn ( x ( n - 1 ) ) ]
##EQU00001##
[0019] wherein, Z_rate is a zero-crossing rate of the voice feature
signal, n is a value among a discrete time series, and
sgn [ x ] = { 1 , x > 0 - 1 , x < 0 } . ##EQU00002##
[0020] Wherein, the hands-free call device further comprises a
receiver end composed of at least one loudspeaker, and the method
further comprises:
[0021] providing a virtual loudspeaker of one loudspeaker in the
receiver end with a connection line between the loudspeaker and its
virtual loudspeaker directed to direction .gamma.1, and defining a
third collection angle directed to a direction determined by the
second collection angle.
[0022] According to another aspect of the present invention, an
apparatus for improving the call quality of a hands-free call
device is provided, comprising:
[0023] a voice feature determination unit configured to scan within
an initial first collection angle of a transmitter end; and after a
voice feature signal is scanned within the first collection angle,
according to a direction of the voice feature signal, determine a
second collection angle smaller than the first collection angle
within the first collection angle; and
[0024] a direction calibration unit configured to calibrate the
transmitter end to a direction determined by the second collection
angle.
[0025] Wherein, the apparatus further comprises: a virtual
microphone creation unit configured to make a reverse extension
line through the main microphone in direction .gamma.1 when a voice
feature signal is scanned in direction .gamma.1 within the initial
first collection angle of the transmitter end, draw a circle by
taking the main microphone as a center and a connection line
between the main microphone and one of the auxiliary microphones as
a radius, and determine intersection of an arc of the circle and
the reverse extension line as a virtual microphone of the main
microphone; and
[0026] an angle determination unit configured to take the main
microphone and its virtual microphone as a new voice array, define
an angle .beta.1 smaller than the first collection angle, and judge
in real time whether a voice feature signal is existed within the
angle .beta.1; if so, determine the angle .beta.1 as a second
collection angle; and if not, take the connection line between the
main microphone and one of the auxiliary microphones as an axis of
symmetry, define a mirror angle .beta.2 of the angle .beta.1 with
respect to the axis of symmetry, and determine the angle .beta.2 as
a second collection angle.
[0027] Wherein, the angle determination unit is further configured
to:
[0028] detect envelope energy of the voice feature signal in
direction .gamma.1, and detect a zero-crossing rate of the voice
feature signal in direction .gamma.1 when an energy detection value
is larger than a first predetermined threshold; and
[0029] determine that a voice feature signal is existed in the
angle .beta.1, when the zero-crossing rate of the voice feature
signal in direction .gamma.1 reaches a second predetermined
threshold.
[0030] Wherein, the angle determination unit comprises:
[0031] an envelope detection unit configured to detect envelope
energy of the voice feature signal in direction .gamma.1 through
the following formula:
power=0;power=power*(1-alpha)+.SIGMA..sub.n=1.sup.Nx(n)*x(n)
[0032] wherein, power is an energy value of the voice feature
signal, parameter alpha is a weighted factor, and parameter N is a
specific value of a voice feature signal at a time point; and
[0033] a zero-crossing detection unit configured to detect a
zero-crossing rate of the voice feature signal in direction
.gamma.1 through the following formula:
Z_rate = 1 2 m = 0 N - 1 | sgn [ x ( n ) - sgn ( x ( n - 1 ) ) ]
##EQU00003##
[0034] wherein, Z_rate is a zero-crossing rate of the voice feature
signal, n is a value among a discrete time series, and
sgn [ x ] = { 1 , x > 0 - 1 , x < 0 } . ##EQU00004##
[0035] Wherein, the apparatus further comprises a receiver end
location unit configured to provide a virtual loudspeaker of one
loudspeaker in the receiver end with a connection line between the
loudspeaker and its virtual loudspeaker directed to direction
.gamma.1, and define a third collection angle having a regional
extent covering a direction determined by the second collection
angle.
[0036] According to another aspect of the present invention, a
hands-free call device is provided, comprising a transmitter end
composed of a main microphone and at least one auxiliary
microphone, a receiver end composed of at least one loudspeaker,
and the aforementioned apparatus for improving the call quality of
the hands-free call device.
[0037] By performing a voice calibration location of the
transmitter end, the method and apparatus for improving the call
quality of a hands-free call device of the present invention narrow
the voice pickup angle of the transmitter end into a relatively
accurate range, thereby avoiding the voice signal to noise ratio
from being influenced by much ambient noise in the voice signal
picked up by the transmitter end, solving the problem that the call
quality of the hands-free call device is not high, and improving
the call quality of the hands-free call device.
[0038] Other characteristics and advantages of the present
application will be elaborated in the subsequent Description, and
they are partly obvious from the Description or acquirable by
implementing the present application. The objective and other
advantages of the present application can be realized and achieved
through the structures particularly pointed out in the Description,
claims and drawings.
BRIEF DESCRIPTION OF THE DRAWINGS
[0039] The drawings are provided for further understanding of the
present invention, and constitute a part of the Description to
explain the present invention together with the embodiments of the
present invention, rather than restricting the present invention.
In which,
[0040] FIG. 1 is a flowchart of a method for improving the call
quality of a hands-free call device provided by one embodiment of
the present invention;
[0041] FIG. 2 is a principle diagram of a method for improving the
call quality of a hands-free call device provided by one embodiment
of the present invention;
[0042] FIG. 3a is a first schematic diagram of a transmitter end
calibration provided by one embodiment of the present
invention;
[0043] FIG. 3b is a second schematic diagram of a transmitter end
calibration provided by one embodiment of the present
invention;
[0044] FIG. 4 is flowchart of determination of a voice feature
signal provided by one embodiment of the present invention;
[0045] FIG. 5 is a schematic diagram of a receiver end calibration
provided by one embodiment of the present invention; and
[0046] FIG. 6 is a block diagram of an apparatus for improving the
call quality of a hands-free call device provided by one embodiment
of the present invention.
DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS
[0047] The core idea of the present invention is to track the
direction of a voice source in real time through a sound source
localization technology for microphone array, and at the same time,
to determine a voice protection angle smaller than an initial voice
protection angle to perform a voice pickup, so that the
interference from surrounding ambient noise can be greatly reduced
to achieve the purpose of improving a transmission signal to noise
ratio. Meanwhile, a directional compensation is made for the
loudspeaker array using orientation information of the voice
source, so that the produced sound is just directed to the voice
source, thereby improving the privacy of reception.
[0048] FIG. 1 is a flowchart of a method for improving the call
quality of a hands-free call device provided by one embodiment of
the present invention. Referring to FIG. 1, the hands-free call
device comprises a transmitter end composed of a main microphone
and at least one auxiliary microphone, and the method
comprises:
[0049] step S110: scanning within an initial first collection angle
of the transmitter end;
[0050] step S120: after a voice feature signal is scanned within
the first collection angle, according to a direction of the voice
feature signal, determining a second collection angle smaller than
the first collection angle within the first collection angle;
and
[0051] step S130: calibrating the transmitter end to a direction
determined by the second collection angle.
[0052] Through those steps, the voice protection angle of the
transmitter end is narrowed to a smaller angle, and a voice pickup
is conducted within the newly determined smaller angle, so that the
interference on the voice signal from ambient noise can be avoided
to improve the transmission signal to noise ratio, thereby
improving the call quality of the hands-free call device.
[0053] FIG. 2 is principle diagram of a method for improving the
call quality of a hands-free call device provided by one embodiment
of the present invention. Referring to FIG. 2, the hands-free call
device 1 comprises: a transmitter end and a receiver end;
[0054] the transmitter end comprises a main microphone MIC-a and an
auxiliary microphone MIC-b, and the receiver end comprises a main
loudspeaker SPK-a and an auxiliary loudspeaker SPK-b;
[0055] an angle .alpha. is an initial scanning angle of the
transmitter end; during a normal communication, the transmitter end
collects a signal, i.e., it firstly scans within a larger
collection angle .alpha., and after a voice feature signal is
detected within the angle .alpha., it narrows the collection angle
to an angle .beta. to achieve the purpose of voice location at the
transmitter end. The process of voice location at the transmitter
end is that the transmitter end scans a voice feature signal within
an initial scanning angle .alpha., and when a voice feature signal
is determined as being existed within an angle .beta., the
transmitter end is located to a direction determined by the angle,
i.e., a position or orientation where the user speaks (as shown in
FIG. 2, the range of the angle .beta. just covers the position of a
head, and the location of the head is more accurate than that made
by the angle .alpha.). The function of the method of the present
invention is to narrow the initial scanning angle to a smaller
second collection angle .beta.. Thus the voice protection angle
becomes smaller, and the position where a user speaks can be
located more accurately, thereby reducing the ambient noise in the
voice signal and improving the voice quality.
[0056] FIG. 3a is a first schematic diagram of a transmitter end
calibration provided by one embodiment of the present invention.
Referring to FIG. 3a, among the reference signs, 1 denotes a
hands-free call device, 2 denotes an auxiliary loudspeaker, 3
denotes an auxiliary microphone, and 4 denotes a virtual microphone
of a main microphone.
[0057] An improvement of the hands-free call device of the present
invention is implemented as follows:
[0058] referring to FIG. 3a, during a communication, firstly a
voice scan is made by the transmitter end within a larger first
collection angle .alpha., and when a voice feature signal is
scanned in direction .gamma.1 within the first collection angle
.alpha., a virtual microphone MIC-c is provided in a reverse
extension line of the main microphone MIC-a from direction
.gamma.1, so as to be located on an arc which takes the main
microphone MIC-a as a center and a connection line between the main
microphone MIC-a and the auxiliary microphone MIC-b as a radius.
Next, the main microphone MIC-a and the virtual microphone MIC-c
constitute a new array, and a directional angle of the new array is
defined as a very small second collection angle .beta.1 directed to
a small range in direction .gamma.1, and the value of .beta.1 is
specifically selected according to different application
scenarios.
[0059] Since the auxiliary array element is absent, with respect to
an array composed of two dot array elements (MIC-a and MIC-c), it
is difficult to distinguish whether a voice comes from the left
side (.gamma.1) or the right side (.gamma.2) of a connection
between the main microphone MIC-a and the auxiliary microphone
MIC-b. One of the core ideas of the method provided by the present
invention is to judge using the virtual microphone MIC-c. The
judgment method firstly provides a virtual microphone MIC-c in
direction .gamma.1 as shown in FIG. 3a and forms a directional
angle .beta.1, so as to judge any voice feature signal within the
directional angle .beta.1 in real time; if a voice feature signal
is found, the directional angle .beta.1 is determined as a second
collection angle of the transmitter end, and the transmitter end is
calibrated to a direction determined by the second collection angle
(.beta.1). If no voice feature signal is found, a virtual
microphone MIC-c is immediately provided anew at an opposite mirror
side of the connection line between the main microphone MIC-a and
the auxiliary microphone MIC-b, so as to judge the direction of the
voice.
[0060] FIG. 3b is a second schematic diagram of a transmitter end
calibration provided by one embodiment of the present invention.
Referring to FIG. 3b, among the reference signs, 1 denotes a
hands-free call device, 2 denotes an auxiliary loudspeaker, 3
denotes an auxiliary microphone, and 4 denotes a virtual microphone
of a main microphone.
[0061] If no voice feature signal is found within the directional
angle .beta.1 as shown in FIG. 3a, a symmetrical angle .beta.2 of
.beta.1 is provided at the other side of the connection between the
main microphone MIC-a and the auxiliary microphone MIC-b as shown
in FIG. 3b, and when a voice feature signal is determined as being
existed within the angle .beta.2, the transmitter end is calibrated
to a direction determined by .beta.2. Through the above steps, the
location and calibration of the transmitter end are completed to
improve the directivity of voice pickup.
[0062] Next, the location of the transmitter end is specifically
described in conjunction with the judgment of the voice feature
signal.
[0063] FIG. 4 is a flowchart of determination of a voice feature
signal provided by one embodiment of the present invention.
Referring to FIG. 4, specifically judging in real time whether any
voice feature signal is existed within the angle .beta.1
comprises:
[0064] 1. collecting a signal, wherein the collected signal is a
scanned voice feature signal in direction .gamma.1;
[0065] 2. detecting envelope energy of the voice feature signal,
and judging whether the energy value is larger than a first
predetermined threshold; if so, detecting a zero-crossing rate of
the voice feature signal; and if not, returning to determine a
direction of the voice feature signal again and a voice feature
signal;
[0066] 3. judging a zero-crossing rate of the voice feature signal;
when the zero-crossing rate is larger than a second predetermined
threshold, determining that a voice feature signal is existed
within the collection angle .beta.1; taking the directional angle
.beta.1 as a smaller second collection angle, so as to calibrate
the transmitter end according to the second collection angle.
[0067] In this embodiment, detecting envelope energy of the voice
feature signal in direction .gamma.1 comprises:
[0068] detecting envelope energy through the following formula:
power=0;power=power*(1-alpha)+.SIGMA..sub.n=1.sup.Nx(n)*x(n)
[0069] wherein, power is an energy value of the voice feature
signal, parameter alpha is a weighted factor, and parameter N is a
specific value of voice feature signal at a time point; wherein the
detection sensitivity is controlled by adjusting the two parameters
alpha and N, so as to ensure stability of the envelope energy
detection. Once the envelope energy power is found to be larger
than the first predetermined threshold (set upon actual
conditions), the step of detecting the zero-crossing rate is
performed.
[0070] Detecting the zero-crossing rate of the voice feature signal
in direction .gamma.1 comprises: detecting the zero-crossing rate
through the following formula:
Z_rate = 1 2 m = 0 N - 1 | sgn [ x ( n ) - sgn ( x ( n - 1 ) ) ]
##EQU00005##
[0071] wherein, Z_rate is a zero-crossing rate of the voice feature
signal, n is a value among a discrete time series, and
sgn [ x ] = { 1 , x > 0 - 1 , x < 0 } . ##EQU00006##
[0072] When the zero-crossing rate Z_rate is larger than the second
predetermined threshold, it is deemed that the collected signals
within the angle .beta.1 contain a voice feature signal, and a
voice action is determined.
[0073] After judging that a voice feature signal is existed within
the angle .beta.1, the transmitter end is located and calibrated to
a range determined by the angle .beta.1. If there is no voice
feature signal within the angle .beta.1 (.gamma.1 is not a voice
source direction), it is judged whether there is any voice feature
signal within an angle .beta.2 symmetrical to .beta.1 about the
connection line between MIC-a and MIC-b. At that time, an envelope
energy detection and a zero-crossing rate detection may be
performed to further verify the voice feature signal within
.beta.2.
[0074] In conclusion, an accurate location of the transmitter end
is achieved by the location of the virtual array element and the
detection of the voice feature signal.
[0075] FIG. 5 is a schematic diagram of a receiver end calibration
provided by one embodiment of the present invention. Among the
reference signs, 1 denotes a hands-free call device, 2 denotes an
auxiliary loudspeaker, and 3 denotes a main microphone.
[0076] After the transmitter end is located, a directional
compensation is made for the loudspeaker array using orientation
information of the voice source, so that the produced sound is just
directed to the voice source, i.e., a sound producing direction of
the loudspeaker array is adjusted, so far as possible, to a
position where the user speaks located by the transmitter end,
i.e., a direction determined by the second collection angle,
thereby improving the privacy of reception. Specifically, a
loudspeaker SPK-b is virtualized using a virtual array element
technology, and a connection line between a virtual loudspeaker
SPK-c and a loudspeaker SPK-a is directed to the voice direction.
Next, a directional angle is provided so that a coverage area of
the third collection angle (i.e., a regional extent of sound
propagation included by the third collection angle) covers a voice
direction determined by the second collection angle (as shown in
FIG. 5, the angle range below SPK-a covers the position of the
head). Preferably, the third collection angle is directly directed
to a voice direction determined by the second collection angle.
[0077] Referring to FIG. 5, the hands-free call device further
comprises a receiver end composed of at least one loudspeaker.
After a calibration of the transmitter end is completed by
determining the second collection angle and direction, the method
further comprises calibrating the receiver end of the hands-free
call device, providing a virtual loudspeaker of one loudspeaker in
the receiver end with the connection line between the loudspeaker
and its virtual loudspeaker directed to direction .gamma.1, and
defining a third collection angle directed to a direction
determined by the second collection angle. The third collection
angle is actually an angle range of sound playing, including the
second collection angle and direction, so that the loudspeaker
plays towards the orientation of the voice source as much as
possible, thereby reducing the voice playing range and improving
the privacy of reception.
[0078] During usage of the hands-free call device, since a distance
between the microphone and the loudspeaker is short, the receiver
end may be calibrated in a direction of the voice feature signal
determined during the calibration of the transmitter end.
[0079] Referring to FIG. 5, a virtual loudspeaker SPK-c of the
auxiliary loudspeaker SPK-b is provided in a reverse extension line
of the main loudspeaker SPK-a from direction .gamma.1, so as to be
located on an arc which takes the main loudspeaker SPK-a as a
center and a connection line between the main loudspeaker SPK-a and
the auxiliary loudspeaker SPK-b as a radius. The main loudspeaker
SPK-a and the virtual loudspeaker SPK-c constitute a new array, and
a directional angle of the new array is defined as a third
collection angle directed to a direction determined by the second
collection angle, which is determined during calibration of the
transmitter end, so as to complete the location and calibration of
the receiver end.
[0080] It shall be appreciated that those angles occurring in FIGS.
3a, 3b and 5 are for the purpose of exemplarily describing the
angle ranges for locating the transmitter end and the receiver end,
and the angular dimensions are not actual. The embodiments of the
present invention improve the signal to noise ratio of the
hands-free voice signal by locating and calibrating the transmitter
end, and improve the privacy of reception by locating the receiver
end.
[0081] In conclusion, by calibrating the transmitter end and the
receiver end, the method for improving the call quality of the
hands-free call device of the present invention reduces the
interference from ambient noise, improves the transmission signal
to noise ratio, and achieves the purpose of improving the call
quality of the hands-free call device. In addition, it performs a
direction compensation for the loudspeaker array, so that the
produced sound is just directed to the voice source, thereby
improving the privacy of reception and the user experience. The
method can be applied to a smart device having the hands-free call
function, such as a smart watch, and greatly improves the call
performance of the smart device.
[0082] According to another aspect of the present invention, an
apparatus for improving the call quality of a hands-free call
device is provided. FIG. 6 is a block diagram of an apparatus for
improving the call quality of a hands-free call device provided by
one embodiment of the present invention, wherein the apparatus 600
comprises:
[0083] a voice feature determination unit 601 configured to scan
within an initial first collection angle of a transmitter end; and
after a voice feature signal is scanned within the first collection
angle, according to a direction of the voice feature signal,
determine a second collection angle smaller than the first
collection angle within the first collection angle; and
[0084] a direction calibration unit 602 configured to calibrate the
transmitter end to a direction determined by the second collection
angle.
[0085] Wherein, the apparatus further comprises a virtual
microphone creation unit configured to make a reverse extension
line through the main microphone in direction .gamma.1 when a voice
feature signal is scanned in direction .gamma.1 within the initial
first collection angle of the transmitter end, draw a circle by
taking the main microphone as a center and a connection line
between the main microphone and one of the auxiliary microphones as
a radius, and determine intersection of an arc of the circle and
the reverse extension line as a virtual microphone of the main
microphone; and
[0086] an angle determination unit configured to take the main
microphone and its virtual microphone as a new voice array, define
an angle .beta.1 smaller than the first collection angle, and judge
in real time whether a voice feature signal is existed within the
angle .beta.1; if so, determine that angle .beta.1 as a second
collection angle; and if not, take a connection line between the
main microphone and one of the auxiliary microphones as an axis of
symmetry, define a mirror angle .beta.2 of the angle .beta.1 with
respect to the axis of symmetry, and determine the angle .beta.2 as
a second collection angle.
[0087] Wherein, the angle determination unit is further configured
to,
[0088] detect envelope energy of the voice feature signal in
direction .gamma.1, and detect a zero-crossing rate of the voice
feature signal in direction .gamma.1 when the energy detection
value is larger than a first predetermined threshold;
[0089] determine that a voice feature signal is existed in the
angle .beta.1, when the zero-crossing rate of the voice feature
signal in direction .gamma.1 reaches a second predetermined
threshold.
[0090] Wherein, the angle determination unit comprises:
[0091] an envelope detection unit configured to detect envelope
energy of the voice feature signal in direction .gamma.1 through
the following formula:
power=0;power=power*(1-alpha)+.SIGMA..sub.n=1.sup.Nx(n)*x(n)
[0092] wherein, power is an energy value of the voice feature
signal, parameter alpha is a weighted factor, and N is a specific
value of the voice feature signal at a time point;
[0093] a zero-crossing detection unit configured to detect a
zero-crossing rate of the voice feature signal in direction
.gamma.1 through the following formula:
Z_rate = 1 2 m = 0 N - 1 | sgn [ x ( n ) - sgn ( x ( n - 1 ) ) ]
##EQU00007##
[0094] wherein, Z_rate is a zero-crossing rate of the voice feature
signal, n is a value among a discrete time series, and
sgn [ x ] = { 1 , x > 0 - 1 , x < 0 } . ##EQU00008##
[0095] Wherein, the apparatus further comprises a receiver end
location unit configured to provide a virtual loudspeaker of one
loudspeaker in the receiver end with a connection line between the
loudspeaker and its virtual loudspeaker directed to direction
.gamma.1, and define a third collection angle having a regional
extent covering a direction determined by the second collection
angle.
[0096] According to another aspect of the present invention, a
hands-free call device is provided, comprising a transmitter end
composed of a main microphone and at least one auxiliary
microphone, a receiver end composed of at least one loudspeaker,
and the aforementioned apparatus for improving the call quality of
the hands-free call device.
[0097] To be noted, the transmitter end and the receiver end of the
hands-free call device of the present invention can be improved in
the aforementioned method for improving the hands-free call
quality. But under some application scenarios, the hands-free call
device only comprises a transmitter end or a receiver end. In that
case, the receiver end may be improved in the method for improving
the hands-free call quality at the receiver end in the embodiment
of the present invention, or the transmitter end may be improved in
the method for improving the hands-free call quality at the
transmitter end in the embodiment of the present invention. That is
to say, the method for improving the hands-free call quality at the
receiver end and at the transmitter end in the present invention
can be implemented separately, which are omitted herein.
[0098] The above descriptions are just preferred embodiments of the
present invention, rather than restrictions to the protection scope
of the present invention. Any amendment, equivalent replacement,
improvement, etc. made within the spirit and principle of the
present invention shall fall within the protection scope of the
present invention.
* * * * *