102 I12 1/14 /
US 20030099345A1
(19)
United States
(12) Patent Application Publication (10) Pub. No.: US 2003/0099345 A1
Gartner et al.
(43) Pub. Date:
May 29, 2003
(54) TELEPHONE HAVING IMPROVED HANDS
FREE OPERATION AUDIO QUALITY AND
Publication Classi?cation
METHOD OF OPERATION THEREOF
(51)
Int. Cl? .
(52)
Us. 01. ............................... .. 379/387.01; 379/388.01
H04M 1/00; H04M 9/00
(75) Inventors: Martin Gartner, Taufkirchen (DE);
Thomas D. Slagle, Boca Raton, FL
(57)
(Us)
Correspondence Address:
Elsa Keller, Legal Assistant
ABSTRACT
A telephone having a hands-free mode of operation. The
telephone includes a pair of microphones spaced apart from
Intellectual Property Department
each other. Each microphone receives sound in hands-free
SIEMENS CORPORATION
186 Wood Avenue South
mode of operation and provides audio signals representative
of received sounds. The audio signals from each microphone
may be converted to digital audio signals. The digital audio
Iselin, NJ 08830 (US)
signals are presented to a ?xed delay path and a variable
(73)
Assignee: Siemens Information
(21)
APPL NO;
09/994,405
?ltered in an adjustable ?lter to remove noise based upon a
prior determination of the noise source location and the
(22)
Filed;
Nov, 27, 2001
voice spectrum derived from the digital audio signals.
102
I12
delay path. Audio signals from both paths are combined and
11B
1/14
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ADC
FIXED
DELAY
128
/
DIGITAL
ADJUSTABLE
AUDIO
OUT
V
126
*
104
118
/
VARIABLE
DELAY
ADC
124
/
130
—~
122
1/90
‘?> PS
-
y
d2
d
ANALYSIS AND CONTROL
no
“
Patent Application Publication
May 29, 2003 Sheet 1 0f 3
US 2003/0099345 A1
FIG. 1
\
/100
10s
3/104
106
FIG. 2
102
112
116
1/14
/
‘
FIXED
ADC
’
DELAY
128
/
126
104
118
122
1/20
ADC
.
VARIABLE
DELAY
/
y
d
130
AUDIO
0m
/
124
‘?> PS
ADJUSTABLE
__ DEILGLILTEiLL ‘
DIGITAL
d
2
ANALYSIS AND CONTROL 1
110
_
Patent Application Publication
May 29, 2003 Sheet 2 0f 3
US 2003/0099345 A1
FIG. 3
142
144\
MICROPHONE
SPACING
NOISE
BASELINE
145\
VOICE
SPECTRUM
1“B\
EXTRACT
DELAYS
150
\
1
SET T2
NOISE
BASELINE
/ 158
Patent Application Publication
May 29, 2003 Sheet 3 0f 3
FIG. 4
,
INCREASE
/1482
US 2003/0099345 A1
May 29, 2003
US 2003/0099345 A1
TELEPHONE HAVING IMPROVED HANDS FREE
OPERATION AUDIO QUALITY AND METHOD OF
OPERATION THEREOF
BACKGROUND OF THE INVENTION
[0001]
1. Field of the Invention
[0002]
The present invention is related to telephones and
more particularly to telephones having a hands-free mode of
BRIEF DESCRIPTION OF THE DRAWINGS
[0011] FIG. 1 shoWs an example of a preferred embodi
ment telephone having a hands-free mode of operation;
[0012] FIG. 2 shoWs a preferred embodiment hands-free
mode circuit for a speakerphone such as the telephone of
FIG. 1;
operation.
[0013] FIG. 3 is a How diagram shoWing steps to set up
and use a preferred embodiment speakerphone;
[0003] 2. Background
[0014]
[0004] Typical state-of-the-art telephones often have a
hands-free or speakerphone mode of operation, hereinafter
generically “speakerphone.” Such a telephone may be
located at a convenient location and placed in hands-free
mode. Thereafter, speakers, e.g., teleconference participants,
may remain stationary or move about Within range of the
speakerphone as desired. The speakerphone microphone
picks up all surrounding sound including background noise.
This sound is transmitted to a listener at the other end of the
call. Traditional speakerphones have a single microphone
and are omnidirectional such that voice of the speaker and
background noise are equally received and passed on to the
listener.
[0005] Occasionally, background noise may be such that
FIG. 4 is an example of how "52 is determined.
DETAILED DESCRIPTION
[0015] FIG. 1 shoWs an example of a preferred embodi
ment telephone 100 With a hands-free mode of operation that
includes a ?rst microphone 102 and a second microphone
104 being used by a speaker 106 in the presence of a noise
source 108. Preferably, the microphones 102, 104 are iden
tical non-directional microphones and are mounted inter
nally to the telephone 100 and spaced as far apart as the
telephone casing alloWs, e.g., in the tWo front corners of the
telephone casing. Thus, a sound from either of speaker 106
or noise source 108 arrives at each of the microphones 102,
104 at slightly different times, normally exhibited as phase
differences. Thus, the dual microphone speakerphone exhib
hands free operation is difficult to use if usable at all. Often
its a directional microphone characteristic When the unde
the background noise originates from a single source that
layed signals from the microphones 102, 104 are combined.
may be located at a ?xed location Within the room, e.g., from
a noisy air conditioner or, from outside of the room such as
[0016]
from street Work. To compensate for this background noise
the microphone sensitivity may be loWered and the speakers
may be requested to speak up. Sometimes this Works,
sometimes it does not. Also, the noise may be such that
setting the microphone sensitivity at one level is an unac
ceptable solution, e.g., a pulsating type noise.
[0006]
In an alternate embodiment the microphones are
external to the speakerphone casing, Wired to the speaker
phone. A larger distance betWeen the tWo microphones
facilitates suppressing the loWer frequency noise sources.
HoWever, this advantage is offset in that large spacing
betWeen the tWo microphones 102, 104 may result in
unequal signal volume betWeen the tWo microphones, espe
cially, if the speaker is much closer to one microphone than
Thus there is a need for a speakerphone With
to the other. Accordingly, this alternate embodiment may
capability of selectively removing background noise to
provide improved audio quality, especially during hands free
signal volume, e.g., one ampli?er, e.g., 118 as shoWn in FIG.
operation.
require additional logic/circuitry to compensate for different
2, may have an adjustable ampli?cation factor.
SUMMARY OF THE INVENTION
[0017] Also, although the present invention is described
noise ratio for telephones operating in hands free mode of
herein as a digital embodiment, this is for example only. The
hands-free telephone of the present invention may be imple
operation;
mented using analog components Without departing from the
[0007]
It is a purpose of the invention to improve a signal
[0008] It is another purpose of the invention to improve
the audio quality provided to a listener at a receiving ends of
a hands free call;
[0009] The present invention is a telephone having a
hands-free mode of operation. The telephone includes a pair
of microphones spaced apart from each other. Each micro
phone receives sound in hands-free mode of operation and
provides audio signals representative of received sounds.
The audio signals from each microphone may be converted
to digital audio signals. The digital audio signals are pre
sented to a ?xed delay path and a variable delay path. Audio
signals from both paths are combined and ?ltered in an
adjustable ?lter to remove noise based upon a prior deter
mination of the noise source location and the voice spectrum
derived from the digital audio signals.
[0010] Additional bene?ts and features of the invention
Will be apparent from the folloWing detailed description
taken together With the attached draWings.
spirit or scope of the invention. Further, directional micro
phones may be substituted for the above described non
directional microphones 102, 104, provided they are
directed toWards the expected speaker location and orthogo
nal to the line de?ned by the microphones.
[0018] For purposes of description of the invention, the
distance betWeen microphones 102 and 104 is referred to
herein as x12. The distance betWeen speaker 106 and micro
phone 102 is referred to herein as xul. The distance betWeen
the speaker 106 and microphone 104 is referred to herein as
xu2. The distance betWeen noise source 108 and microphone
102 is referred to herein as xnl. The distance betWeen noise
source 108 and microphone 104 is referred to as xn2.
Although, it is understood that the speed of sound varies
With media and ambient conditions, for the purposes of this
invention and, because normal operating conditions of a
speakerphone for such a conference call are approximately
constant,
Thus,
the the
delay
speed
'51 betWeen
of soundtheis tWo
treated
microphones
as a constant
is deter
May 29, 2003
US 2003/0099345 A1
mined by x12 divided by c, i.e., t1=x12/c. Noise originating
varies betWeen constructive and destructive interference,
at noise source 108 in FIG. 1 arrives at microphones 102,
104 at times offset by (xn1—xn2)/c. Sound from a speaker
106 arrives at microphones 102, 104 at times offset by
While the desired signals (xul, xu2) from the speaker or
speakers alWays add constructively to provide a positive
audio component. Taking the analog sound signal from
(xu1—xu2)/c.
microphones 102, 104 to be X1, X2, respectively, d1=X1
[0019] In the above alternate embodiment Wherein micro
phones 102, 104 are external, '51 may be derived directly. A
(i+'c1) and d2=X2(i), Where
is the digital value of X at
time i. Analysis and Control Unit 110 delays X2(i) betWeen
0 and 251, ?rst to identify the delay to minimiZe noise during
tone may be radiated from one of the tWo microphones, e.g.,
102. The delay betWeen When the tone originates at the ?rst
microphone 102 and When it is received at the second
microphone 104 is '51.
[0020] FIG. 2 shoWs a preferred embodiment hands-free
mode circuit 110 for a speakerphone such as telephone 100
of FIG. 1. Sound signals from one microphone 102 pass
through a ?xed delay path that includes an input ampli?er
baseline determination and second to determine the delay to
maximiZe xu/xn during voice spectrum analysis. Also, voice
spectrum analysis results are applied to Adjustable Digital
Filter 128 to enhance frequencies originating primarily from
the speaker, and to dampen frequencies that originate pri
marily or solely from the noise source 108. Therefore, as
described hereinbeloW, each of these frequency bands are
identi?ed in one of tWo different learning phases. In a ?rst
112, Analog-to-Digital Converter (ADC) 114 and ?xed
delay 116. Coincidentally, sound signals from the second
microphone 104 pass through a variable delay path that
idle-state phase, the typical noise source spectrum is deter
mined to identify the noise frequency bands. Then, in a
includes an input ampli?er 118, an ADC 120 and an adjust
mode and the composite sound that includes both noise and
able variable delay 122. The outputs of ?xed delay 116 and
the speaker’s voice is analyZed to determine the speaker’s
variable delay 122 are combined in adder 126. The outputs
of ADC 120 and ?xed delay 116 also are passed as inputs to
Analysis and Control unit 124. The output of adder 126 is
[0024] Accordingly, having thus characteriZed the circuit
passed to Adjustable Digital Filter 128. Analysis and Control
unit 124 provides control for both adjustable variable delay
122 and Adjustable Digital Filter 128. Adjustable Digital
Filter 128 provides a digital audio output that is the audio
signal passed to a listener at the other end of the call. Phone
status signals 130 are passed as inputs to Analysis and
Control unit 124.
[0021]
The ampli?ers 112, 118 of each path act as a
speaker phase, the speakerphone is placed in hands-free
frequency spectrum.
response to both speaker input and noise input, the circuit
may be calibrated to ?lter out noise. While it is preferred that
the ampli?ers 112, 118 as Well as the ADCs 114, 120 are
identical, in practice some slight differences alWays exist.
These variations in or, differences betWeen components in
each of the paths may be compensated, preferably, during
factory calibration, e. g., by adjusting the ampli?cation factor
of either or both of the ampli?ers 112, 118. By selectively
adjusting variable delay 122 it is possible to folloW the
preampli?er to amplify the sound signal from the particular
connected microphone 102, 104. The output of ampli?ers
speaker’s voice as the speaker moves about the set of
112, 118 are each passed to a respective ADC 114, 120. The
ADCs 114, 120 convert the analog outputs from the corre
directional microphone automatically to the user. As the
variable delay 122 is changed to compensate or to coordi
sponding ampli?ers 112, 118 to a digital output. The digital
nate With changes of speaker location, background noise,
Which originates elseWhere, is dampened or, possibly,
removed. The degree of dampening for the background
noise depends upon its angle of origin and Wavelength in
output signal from ADC 114 is passed to a ?xed delay 116.
Fixed delay 116 is set at '51 (i.e., x12/C). The digital output
from ADC 120 is passed to adjustable variable delay 122.
The Analysis and Control unit 124 may be a simple embed
ded processor or microcontroller (not shoWn) and appropri
ate program code, e.g., stored in a local read only memory
(ROM) or electrically programmable ROM (EPROM). The
Analysis and Control unit 124 controls delay in variable
delay 122 and sets the ?lter bandWidth of Adjustable Digital
Filter 128. Variable delay 122 has an adjustable delay of 62
that may be adjusted to values ranging betWeen 0 and 251.
[0022] In yet another alternate embodiment, both delays
116, 122 are adjustable variable delays, having a range
betWeen 0 and '51. This alternate embodiment maintains
overall circuit delay at a minimum. Accordingly, for this
alternate embodiment, Analysis and Control unit 124 pro
vides control to both adjustable delays.
[0023] Microphone input signals from microphone 102
microphones 102, 104. This is analogous to pointing a single
relation to the noise source distance from the microphones
102, 104, i.e., loWer frequency sound (sub 100 HZ) tends to
be non-directional. Since the loWer the frequency (f), the
longer the Wavelength (8), loWer frequency sound is less
subject to positional ?ltering and dampening. HoWever, such
loW frequency noise may be removed With a simple loW pass
?lter or its equivalent in Adjustable Digital Filter 128.
[0025] FIG. 3 is a How diagram 140 shoWing set up and
use of a preferred embodiment such as speakerphone 100 of
FIG. 1. First, in step 142 the spacing betWeen the micro
phones is input to determine "c1, e.g., entering the ?xed delay
betWeen internal microphones 102, 104 at the factory or, for
the above described external microphone embodiment, auto
matically measuring the delay betWeen origination and
reception of a tone. Then, in step 144 the background noise
is checked. Typically, this check is done When the phone is
(d1) and from microphone 104 (d2) are added constructively
idle such as prior to making a call, at the beginning of a
by setting "c2=t1—(xu2—xu1)/c, Which is maximum (261)
conference call, etc. So, in this step 144 the phone is placed
When the noise source is colinear With the microphones and
separated from microphone 102 by microphone 104, i.e.,
in hands free mode and silence is maintained to generate a
noise baseline With any noise sources that happen to be
microphone 104 is betWeen noise source 108 and micro
phone 102. Thus, for the above described range of '52, the
signals at the tWo microphones 102, 104 may be added to
produce a result Wherein the resulting noise component
Within range of the phone. Next, in step 146, a second
learning or voice baseline step, the speakerphone operates in
hands-free mode and a speaker speaks from Within range of
the phone to obtain a voice spectrum signal. The Analysis
May 29, 2003
US 2003/0099345 A1
and Control unit 124 processes the signals from both micro
phones to extract the voice spectrum from the background
sounds using the background noise information obtained in
step 144. The Adjustable Digital Filter 128 is adjusted to
selectively enhance speech and suppress the background
sounds.
[0026] So, in step 148 the Analysis and Control Unit 124
extracts delays both for noise sources and for voices as
described hereinbeloW With reference to FIG. 4. In step 150,
the optimum delay to maximiZe the voice to noise signal
ratio (xu/xn) is set for '52, the adjustable variable delay 122
in the path from microphone 104. The path outputs from
?xed delay 116 and variable delay 122 are combined in
adder 126 and that sum is passed to the adjustable digital
?lter 128. In step 152 the adjustable digital ?lter is adjusted
to maximiZe speech and, simultaneously, suppress noise
With the ?ltered result being passed to called parties. As long
as the call continues in step 154 and While the speaker is
speaking in step 156, this variable delay calibration may be
repeated, periodically, in step 148 to folloW the speaker.
Also, in step 156 When the Analysis and Control Unit 110
determines that no one is speaking, noise from the noise
source may be re-analyZed in step 158 and the variable delay
calibration repeated in step 148. When hands-free mode
ends or the call ends in step 154, the ?ltering ends in step
160.
[0027]
FIG. 4 shoWs an example of how "52 may be
determined in step 148. Essentially, in each pass through
step 148, "52 is varied slightly (slightly increased/decreased)
and, then, the speaker’s voice to noise signal ratio (xu/xn) is
checked until the optimum delay is found for '52, i.e., Where
any change in "52 reduces xu/xn. Adjustable variable delay
122 is then set to the optimum value of "52 in step 150. During
the initial pass through step 148, T2=T1 and xu/xn is marked
or noted. Thereafter, in step 1482 the delay value for "52 is
increased slightly and in step 1484, xu/xn is checked to
determine if it has increased. If xu/xn increases in step 1484
an optimum value has not yet been identi?ed and, returning
to step 1482, "52 is increased again. Iteratively increasing "c2
and checking xu/xn in steps 1482, 1484 continues until "52 is
maximum (251) or, xu/xn is not found to have increased in
step 1484. If xu/xn decreases after the ?rst increase of "52 in
step 1482 xu/xn is not optimum. OtherWise When xu/xn
decreases, the optimum value of xu/xn has been found in
step 1486 (i.e., one increment beloW the current value) and
in step 1488, "52 is backed off one increment (unless it is at
its maximum value) and that value is passed to step 150.
[0028] If xu/xn decreases after the ?rst increase, then the
optimum value for "52 has not been found in step 1486. So,
the optimum value lies beloW the current value and in step
1490, the delay value for "52 is decreased slightly and in step
1492 xu/xn is checked to determine if it has increased. Steps
1490, 1492 are repeated iteratively, decreasing "c2 and check
ing xu/xn until "52 is minimum (0) or xu/xn is not found to
have increased in step 1492. Again in step 1488, "52 is backed
off one increment (unless it is at its minimum value) and that
value is passed to step 150.
[0029] Thus, the results of the analysis in the learning
steps 144, 146 are combined to automatically maximiZe
xu/xn and provide an optimal ?lter for the hands free phone.
The result favors voice based signals over background noise.
[0030] Accordingly, the dual microphone hands free tele
phone provides a microphone characteristic that is superior
to single microphone telephones, While using a non-me
chanical, dynamically adjustable reception direction. The
background and voice analysis as described for FIG. 3
provides an optimal ?lter for the dual microphone telephone.
In particular analysis is simple enough that recalibration
may be done periodically, manually or automatically
throughout the call to identify background noise. The back
ground noise may be analyZed While the telephone is idle or
during hands free operation, if no one is speaking. The
digital audio output may be provided to any typical tele
phone equipment, e.g., converting the ?ltered digital audio
back to an analog signal for analog transmission or, sending
it as voice over internet protocol (VoIP).
[0031] Thus, the dual microphone telephone of the present
invention provides a signi?cant audio quality improvement
during hands free operation over prior art bands free tele
phones. Further, automatic recalibration may not require
users to perform additional tasks or, at most, may require
performing minimal additional tasks, e.g., initiating each of
the learning steps.
[0032] While the invention has been described in terms of
preferred embodiments, those skilled in the art Will recog
niZe that the invention can be practiced With modi?cation
Within the spirit and scope of the appended claims.
What is claimed is:
1. A telephone having a hands-free mode of operation,
said telephone comprising:
a ?rst microphone receiving sound in hands-free mode,
and providing ?rst audio signals representative of
received sounds to a ?rst delay path;
a second microphone receiving said sounds in hands-free
mode and providing second audio signals representa
tive of said received sounds to a second delay path, said
second microphone spaced a selected distance from
said ?rst microphone;
an adder combining said ?rst audio signals from said ?rst
delay path With said second audio signals from said
second delay path;
an analysis and control unit analyZing received said
signals and adjusting delay through said second delay
path; and
an adjustable ?lter receiving combined said signals from
said adder and ?ltering noise from said combined
signals.
2. A telephone as in claim 1 Wherein said ?rst delay path
is a ?xed delay path, said second delay path is a variable
delay path and said ?rst audio signals from said ?xed delay
path and said second audio signals from said variable delay
path are digital audio signals.
3. A telephone as in claim 2 Wherein said ?xed delay path
provides a delay proportional to the selected distance
betWeen said ?rst microphone and said second microphone.
4. The telephone as in claim 2 Wherein the variable delay
inserts a delay having a range less than tWice the delay of
said ?xed delay path.
5. A telephone as in claim 1 Wherein said adjustable ?lter
is an adjustable digital ?lter providing a digital audio output.
6. A telephone as in claim 2 Wherein each of said ?xed
delay path and said variable delay path comprises:
May 29, 2003
US 2003/0099345 A1
an ampli?er receiving an analog signal from a connected
microphone; and
an analog-to-digital converter (ADC) converting an out
put of said ampli?er to a corresponding digital signal.
7. A telephone as in claim 6 Wherein said digital signal
from said ADC in said variable delay path is provided to said
analysis and control unit.
8. Atelephone as in claim 2 Wherein said output from said
?xed delay path is provided to said analysis and control unit.
9. A telephone as in claim 2 Wherein said analysis and
control unit further sets ?lter values in said adjustable digital
?lter.
10. A telephone as in claim 2 Wherein the analysis and
control unit comprises:
means for varying the delay of said adjustable delay path;
b) taking a voice baseline at each of said microphones,
said voice baseline providing a voice frequency spec
trum of a speaker’s voice;
c) comparing said voice baseline With said noise baseline
to determine a substantially optimum delay for a signal
path from one of said microphones;
d) setting a delay in said signal path responsive to said
optimum delay; and
e) ?ltering noise associated With said noise spectrum.
14. A method as in claim 13 Wherein steps c) through e)
are periodically repeated throughout a hands-free call.
15. Amethod as in claim 14 Wherein at least one idle time
is identi?ed in said hands-free call and, at each said at least
one idle time a neW noise baseline is extracted from signals
determining means for determining a ratio of a voice
signal to a background noise signal; and
means for identifying an increase in said ratio responsive
to delay changes in said adjustable delay path.
11. A telephone as in claim 10 Wherein said analysis and
control unit further comprises:
means for extracting a noise spectrum from a ?rst signal;
and
means for extracting a voice spectrum from a composite
signal, extracted said voice spectrum being compared
from said microphones.
16. A method as in claim 13 Wherein the step c) of
comparing said voice baseline With said noise baseline
comprises the steps of:
i) incrementally increasing said delay;
ii) comparing a voice to noise signal ratio at said increased
delay With a previous voice to noise signal ratio to
determine if said voice to noise signal ratio is
increased; and,
against said noise spectrum in said determining means.
12. A telephone as in claim 11 Wherein said adjustable
iii) repeating steps i) and ii) until said voice to noise signal
digital ?lter is an adjustable bandpass ?lter and said analysis
and control unit adjusts said adjustable bandpass ?lter to
remove signals having frequencies outside of said extracted
voice spectrum.
17. A method as in claim 13 Wherein the step c) of
comparing said voice baseline With said noise baseline
13. A method of controlling a speakerphone, said speak
erphone having at least tWo microphones spaced a selected
distance from each other, sound signals from each of said
microphones being combined in said speaker phone and
presented as a voice output from said speakerphone to a
party at another end of a hands-free call, said method
comprising the steps of:
a) taking a noise baseline at each of said microphones,
said noise baseline providing a noise frequency spec
trum of background noise;
ratio is determined not to have increased.
comprises the steps of:
i) incrementally decreasing said delay;
ii) comparing a voice to noise signal ratio at said increased
delay With a previous voice to noise signal ratio to
determine if said voice to noise signal ratio is
increased; and,
iii) repeating steps i) and ii) until said voice to noise signal
ratio is determined not to have increased.
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