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Claims  |
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What is claimed is:
1. An apparatus for discriminating between signals produced by a desired
acoustic source and an undesired acoustic source, the apparatus
comprising:
a first microphone disposed at a first distance from the desired acoustic
source;
at least a second microphone disposed at a second distance from the desired
acoustic source;
a proximity estimation block configured to utilize signals from the first
microphone and second microphone to produce a signal representing an
estimate of the proximity of the desired acoustic source based upon a
ratio of signals produced by the first microphone and the second
microphone; and
a variable gain block configured to adjust the gain of the output signal
from at least one of the microphones by utilizing the estimate of the
proximity of the desired acoustic source.
2. The apparatus of claim 1 wherein the output signal from the first
microphone is amplified by a gain amount dependent on the magnitude of the
estimate of the proximity.
3. The apparatus of claim 1 wherein the output signals from the microphones
are filtered in the frequency domain prior to estimation of the proximity.
4. The apparatus of claim 1 wherein the ratio is used as an indicator of
speech activity for use in a subsequent processing stage.
5. The apparatus of claim 4 wherein the subsequent processing stage is
configured to suppress side tones in response to a predetermined value of
the estimate of proximity.
6. The apparatus of claim 4 wherein the subsequent processing stage
includes a speech recognition circuit that is enabled in response to a
predetermined value of the estimate of proximity.
7. The apparatus of claim 1 wherein the amplification provided to the
output of the first microphone is a linear function of the ratio.
8. The apparatus of claim 1 wherein the amplification provided to the
output of the first microphone is a logarithmic function of the ratio.
9. The apparatus of claim 1 wherein the amplification provided to the
output of the first microphone is a non-linear function of the ratio.
10. The apparatus of claim 1 wherein the proximity estimation block
includes a divider for receiving and processing the output signals from
the microphones to generate the signal representing an estimate of the
proximity of the desired acoustic sources.
11. The apparatus of claim 10 wherein the divider is configured to take the
difference of the logarithm of the output signals from the microphones.
12. The apparatus of claim 10 wherein the divider includes an anti-log
amplifier for generating an antilog value of the difference of the
logarithm of the output signals from the microphones.
13. The apparatus of claim 10 further comprising a pair of detectors
coupled to input terminals of the divider and configured to permit the
divider to operate in a single quadrant.
14. The apparatus of claim 10 further comprising a pair of band-pass
filters coupled to input terminals of the divider and configured to pass
frequencies occurring within a predetermined range.
15. The apparatus of claim 10 further comprising a clipping and smoothing
block coupled to an output terminal of the divider and configured to clip
and filter the output signal of the divider.
16. The apparatus of claim 1 further comprising a delay stage for
compensating for delays in the proximity estimation before amplification
of the output of the first microphone.
17. The apparatus of claim 1 wherein the proximity estimation block
comprises a digital signal processor for producing a signal representing
an estimate of the proximity of the desired acoustic source.
18. A method of discriminating between signals produced by a desired
acoustic source and an undesired acoustic source, comprising:
using a first microphone disposed at a distance from the desired acoustic
source;
using a second microphone disposed at a second distance from the desired
acoustic source;
utilizing signals from the first microphone and second microphone to
produce an output signal representing an estimate of the proximity of the
desired acoustic source based upon a ratio of signals produced by the
first microphone and the second microphone; and
adjusting the gain of the signal from at least one of the first and second
microphones by utilizing the estimate of the proximity of the desired
acoustic source.
19. The method of claim 18 wherein the signal from the first microphone is
amplified in response to the estimate of the proximity exceeding a
threshold value.
20. The method of claim 18 wherein the signal from the first microphone is
amplified by a gain amount dependent upon the magnitude of the estimate of
the proximity.
21. The method of claim 18 further comprising:
compensating for delays prior to amplification of the output signal from
the first microphone to correct for delays due to the estimation of the
proximity.
22. The method of claim 18 wherein the output signals from the microphones
are filtered in the frequency domain prior to estimating the proximity.
23. The method of claim 18 further comprising:
using the estimate of proximity as an indicator of speech activity for use
in a subsequent processing stage.
24. The method of claim 18 wherein the amplification provided to the output
of the first microphone is a linear function of the estimate of proximity.
25. The method of claim 18 wherein the amplification provided to the output
of the first microphone is a logarithmic function of the estimate of
proximity.
26. The method of claim 18 wherein the amplification provided to the output
of the first microphone is a non-linear function of the estimate of
proximity.
27. An apparatus for discriminating between signals produced by a desired
acoustic source and an undesired acoustic source, the apparatus
comprising:
an input block including a first microphone disposed at a first distance
from the desired acoustic source and a second microphone disposed at a
second distance from the desired acoustic source;
a proximity estimation block coupled to the input block and configured to
produce an estimate of the proximity of the desired acoustic source based
upon output signals of the microphones;
a speech activity detector coupled to the proximity estimation block and
configured to produce a control signal in response to the estimate of the
proximity;
a differential amplifier having inputs coupled to the first microphone and
the second microphone and having an output;
a variable gain block coupled to the output of the differential amplifier
and the proximity estimation block, and configured to adjust the gain of
the output signal from the differential amplifier based upon the estimate
of the proximity; and
a microphone adjustment stage coupled to each of the microphones, the
microphone adjustment stage permitting the sensitivities of the
microphones to be adjusted and substantially matched.
28. The apparatus of claim 27 wherein the estimate of the proximity is
based upon the ratio of the output signals produced by the microphones.
29. The apparatus of claim 27 wherein the speech activity detector
comprises a comparator having a first input coupled to the proximity
estimation block, a second input coupled to a proximity threshold
reference source, and an output for producing an speech activity output
signal.
30. The apparatus of claim 27 further comprising a delay stage coupled
between the first microphone and the variable gain block, and configured
to compensate for delays provided by the proximity estimation block.
31. The apparatus of claim 27 wherein the amplification provided to the
output of the first microphone is a linear function of the estimate of the
proximity.
32. The apparatus of claim 27 wherein the amplification provided to the
output of the first microphone is a non-linear function of the estimate of
proximity.
33. The apparatus of claim 27 wherein the proximity estimation block
comprises a digital signal processor for producing the estimate of the
proximity.
34. An apparatus for discriminating between signals produced by a desired
acoustic source and an undesired acoustic source, the apparatus
comprising:
a first microphone disposed at a first distance from the desired acoustic
source;
at least a second microphone disposed at a second distance from the desired
acoustic source;
a proximity estimation block configured to utilize signals from the first
microphone and second microphone to produce a signal representing an
estimate of the proximity of the desired acoustic source;
a variable gain block configured to adjust the gain of an output signal
from at least one of the microphones by utilizing the estimate of the
proximity of the desired acoustic source; and
a speech activity detector coupled to the proximity estimation block and to
the variable gain block, and configured to indicate the presence of
desired signals from the desired acoustic source.
35. An apparatus for discriminating between signals produced by a desired
acoustic source and an undesired acoustic source, the apparatus
comprising:
an input block including a first microphone disposed at a first distance
from the desired acoustic source and a second microphone disposed at a
second distance from the desired acoustic source;
a proximity estimation block coupled to the input block and configured to
produce an estimate of the proximity of the desired acoustic source based
upon output signals of the microphones;
a speech activity detector coupled to the proximity estimation block and
configured to produce a control signal in response to the estimate of the
proximity; and
a variable gain block coupled to the input block and the speech activity
detector, and configured to adjust the gain of the output signal from at
least one of the microphones based upon the estimate of the proximity.
36. The apparatus of claim 35 wherein the estimate of the proximity is
based upon the ratio of the output signals produced by the microphones.
37. The apparatus of claim 35 wherein the speech activity detector
comprises a comparator having a first input coupled to the proximity
estimation block, a second input coupled to a proximity threshold
reference source, and an output coupled to the variable gain block.
38. The apparatus of claim 35 further comprising a delay stage coupled
between the first microphone and the variable gain block, and configured
to compensate for delays provided by the proximity estimation block.
39. The apparatus of claim 35 wherein the proximity estimation block
comprises a digital signal processor for producing the estimate of the
proximity.
40. An apparatus for discriminating between signals produced by a desired
acoustic source and an undesired acoustic source, the apparatus
comprising:
a first microphone disposed at a first distance from the desired acoustic
source;
at least a second microphone disposed at a second distance from the desired
acoustic source;
a proximity estimation block configured to utilize signals from the first
microphone and second microphone to produce a signal representing an
estimate of the proximity of the desired acoustic source; and
a variable gain block configured to adjust the gain of an output signal
from at least one of the microphones by utilizing the estimate of the
proximity of the desired acoustic source, wherein the gain applied to the
output signal from the first microphone is determined by both the level of
output signal and by the estimate of the proximity exceeding a threshold
value.
41. The apparatus of claim 40 wherein the output signal from the first
microphone is amplified in response to the estimate of the proximity
exceeding the threshold value.
42. A method of discriminating between signals produced by a desired
acoustic source and an undesired acoustic source, comprising:
using a first microphone disposed at a distance from the desired acoustic
source;
using a second microphone disposed at a second distance from the desired
acoustic source;
utilizing signals from the first microphone and second microphone to
produce an output signal representing an estimate of the proximity of the
desired acoustic source; and
adjusting the gain of the signal from at least one of the first and second
microphones by utilizing the estimate of the proximity of the desired
acoustic source wherein the gain applied to the signal from the first
microphone is determined by a level of the signal and by the estimate of
the proximity exceeding a threshold value.
43. An apparatus for discriminating between signals produced by a desired
acoustic source and an undesired acoustic source, the apparatus
comprising:
an input block including a first microphone disposed at a first distance
from the desired acoustic source and a second microphone disposed at a
second distance from the desired acoustic source;
a proximity estimation block coupled to the input block and configured to
produce an estimate of the proximity of the desired acoustic source based
upon output signals of the microphones;
a speech activity detector coupled to the proximity estimation block and
configured to produce a control signal in response to the estimate of the
proximity; and
a variable gain block coupled to the input block and the proximity
estimation block, and configured to adjust the gain of the output signal
from at least one of the microphones based upon the estimate of the
proximity.
44. The apparatus of claim 43 wherein the estimate of the proximity is
based upon the ratio of the output signals produced by the microphones.
45. The apparatus of claim 43 wherein the speech activity detector
comprises a comparator having a first input coupled to the proximity
estimation block, a second input coupled to a proximity threshold
reference source, and an output for producing an speech activity output
signal.
46. The apparatus of claim 43 further comprising a delay stage coupled
between the first microphone and the variable gain block, and configured
to compensate for delays provided by the proximity estimation block.
47. The apparatus of claim 43 wherein the amplification provided to the
output of the first microphone is a linear function of the estimate of the
proximity.
48. The apparatus of claim 43 wherein the amplification provided to the
output of the first microphone is a non-linear function of the estimate of
proximity.
49. The apparatus of claim 43 wherein the proximity estimation block
comprises a digital signal processor for producing the estimate of the
proximity.
50. A method of discriminating between signals produced by a desired
acoustic source and at least one undesired acoustic source, the method
comprising:
using a first microphone disposed at a distance from the desired acoustic
source;
using a second microphone disposed at a greater distance from the desired
acoustic source and bearing a known relationship to the first microphone;
adjusting sensitivities of the first and second microphones to be
substantially matched;
utilizing the output signals from the first microphone and the second
microphone to produce a signal representing an estimate of the proximity
of the desired acoustic source in an active state;
producing an output based on the difference between the instantaneous
values of the output signals from the first microphone and the second
microphone in order to produce a composite output with the properties of a
directional microphone; and
adjusting a gain of the output based on the difference between the
instantaneous values of the output signals from the first microphone and
the second microphone by use of the estimate of the proximity of the
desired acoustic source.
51. The method of claim 50 further comprising:
filtering in the frequency domain the output signals from the first
microphone and the second microphone prior to the estimation of the
proximity.
52. The method of claim 53 further comprising:
digitizing the output signals of the first microphone and the second
microphone prior to processing the output signals.
53. An apparatus for discriminating between signals produced by a desired
acoustic source and an undesired acoustic source, comprising:
a first microphone disposed at a distance from the desired acoustic source;
a second microphone disposed at a second distance from the desired acoustic
source;
means, coupled to the first microphone and second microphone, for utilizing
signals from the first microphone and second microphone to produce an
output signal representing an estimate of the proximity of the desired
acoustic source based upon a ratio of signals produced by the first
microphone and the second microphone; and
means, coupled to the utilizing means, for adjusting the gain of the signal
from at least one of the first and second microphones by utilizing the
estimate of the proximity of the desired acoustic source.
54. An apparatus for discriminating between signals produced by a desired
acoustic source and at least one undesired acoustic source, the method
comprising:
a first microphone disposed at a distance from the desired acoustic source;
a second microphone disposed at a greater distance from the desired
acoustic source and bearing a known relationship to the first microphone;
means, coupled to the first microphone and second microphone, for utilizing
the output signals from the first microphone and the second microphone to
produce a signal representing an estimate of the proximity of the desired
acoustic source in an active state;
means, coupled to each of the microphones, for permitting sensitivities of
the microphones to be adjusted and substantially matched;
means, coupled to the first microphone and second microphone, for producing
an output based on the difference between the instantaneous values of the
output signals from the first microphone and the second microphone in
order to produce a composite output with the properties of a directional
microphone;
means, coupled to the utilizing means, for adjusting a gain of the output
based on the difference between the instantaneous values of the output
signals from the first microphone and the second microphone by use of the
estimate of the proximity of the desired acoustic source. |
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Claims |
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Description |
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FIELD OF THE INVENTION
The present invention relates generally to the field of communications with
possible uses in voice recognition, public address, and recording. The
present invention relates more particularly to a signal expander system
that can distinguish sounds from acoustic sources placed at different
locations relative to a transducer.
BACKGROUND OF THE INVENTION
Conventional voice communication systems typically incorporate some form of
voice activated switch or signal expander for suppressing acoustic
background noises. FIG. 1A illustrates a functional block diagram of a
conventional bi-stable signal expander system 100 comprising a microphone
105, an expander control stage 110, and a variable gain block (stage) 115.
The microphone 105 is coupled to the variable gain block 115, and the
expander control stage 110 is coupled to the microphone 105 and the
variable gain block 115. The expander control stage 110 includes a
detector 120 coupled between the microphone 105 and a first input 125 of a
comparator 130. The comparator 130 has a second input 135 coupled to a
reference (threshold) voltage source 140 for generating a reference
voltage level Vref.
When an acoustic source 145 becomes active, the emitted sounds from
acoustic source 145 will cause changes in the air pressure. The microphone
105 detects the air pressure changes and translates the air pressure
changes into corresponding voltage changes (i.e., microphone output
signals 150) that are detected by the detector 120. The detector 120
outputs the microphone output signal 150 as a detector output signal 155.
The comparator 130 compares the voltage level of the detector output
signal 155 with the reference voltage level Vref from reference voltage
source 140. If the voltage level of the detector output signal 155 is
below the reference voltage level Vref, then the comparator 130 generates
an output signal 160 with a logical state that does not activate the
variable gain block 115. As a result, the variable gain block 115 does not
add gain to the microphone output signal 150. When the acoustic source 145
activates, the voltage level of the microphone output signal 150 rises.
The detector 120 detects the higher-level microphone output signal 150 and
will, as a result, output a higher-level detector output signal 155. If
the voltage level of detector output signal 155 rises above the reference
voltage level Vref, the comparator 130 outputs a control signal 160 with a
logic state that causes the variable gain block 115 to add gain to the
microphone output signal 150. The variable gain block 115 then outputs the
amplified microphone output signal as an audio output signal 165.
A disadvantage of the bi-stable signal expander system 100 is that a
low-level sound (e.g., soft speech) from the acoustic source 145 may not
be amplified if the magnitude of the low level sound does not trigger the
comparator 130. If the threshold of comparator 130 is set too low, then
noise signals in the environment will easily trigger the comparator 130,
thereby activating the variable gain block 115 and amplifying the noise
signals ("noise pumping"). If the threshold of comparator 130 is set too
high, then softer sounds from the acoustic source 145 may not trigger the
comparator 130 and not activate the variable gain block 115, resulting in
inadequate gain for the desired signal.
In order to reduce the consequence of low level speech failing to activate
the comparator 130, some systems set the minimum gain of the variable gain
block 115 to be only 12 dB to 20 dB below the maximum (fully activated)
gain.
A disadvantage of the above conventional bi-stable signal expander systems
is that ambient or undesired noises having magnitudes above the threshold
level of comparator 130 are amplified. Additionally, if the bi-stable
expander system 100 will be used in a noisy environment, the threshold of
the comparator 130 must be set appropriately so that the external noises
do not trigger the comparator 130, increasing the severity of noise
modulation and increasing the likelihood that the system will not respond
properly to voice.
Additionally, if the microphone 105 is oriented away from the acoustic or
speech source 145, the microphone 105 may not be able to properly detect
the desired sound waves. As a result, the microphone output voltage level
150 will be low and may not trigger the comparator 130 to permit
amplification of the desired detected sound.
The above-mentioned bi-stable signal expander systems have a fast attack
and slow decay characteristic that causes the switches for controlling
gain to respond quickly to a detected sound of a sufficient voltage level
and to maintain the gain for a pre-defined time length (e.g., 150 ms to
200 ms) after the voltage level of the detected sound falls below the
comparator 130 threshold. By maintaining the gain for the additional
pre-defined time length, the quieter-sounding, trailing ends of the speech
envelope are not cut off by the bi-stable signal expander system. These
trailing ends are typically below the comparator threshold. However, noise
is often amplified during the additional pre-defined time length when gain
is maintained.
The above-mentioned bi-stable signal expander systems also encounter
problems when a burst of background noise occurs. For example, the noise
burst might be a typewriter key impact or other types of noises with
impulse waveforms. The noise burst will trigger the comparator 130 in the
bi-stable signal expander system, thereby adding gain to the undesired
noise burst. In addition, since the gain is maintained for the
above-mentioned pre-defined time length after the noise burst occurrence,
subsequent undesired noises are also amplified.
FIG. 1B illustrates a conventional variable gain signal expander system 200
including an expander control stage 205 coupled between the microphone 105
and the variable gain block 115. The expander control stage 205 includes a
detector 210 coupled to an amplifier 215. The microphone output signal 150
is detected by the detector 210 and amplified by the amplifier 215. As a
result, the amplifier 215 generates a control signal 220 with a magnitude
that depends on the initial magnitude of the microphone output signal 150.
The amount of gain provided by the variable gain block 115 to the
microphone output signal 150 depends on the magnitude of the control
signal 220.
The variable gain signal expander system 200 can be designed with a shorter
time constant for reduced audibility of "noise pumping". The shorter time
constant reduces the amount of time that high gain is applied to the noise
signal as the desired signal drops in amplitude. The effect of the
combination of variable gain with reduced time constants on the desired
signal is to modulate the envelope of the speech signal. This is not
generally desirable but may be an acceptable compromise in noisy
environments.
A further disadvantage of the variable gain signal expander system 200 is
that both the signal from the desired acoustic source and the ambient
acoustic noise are detected and used to increase the gain of the variable
gain block 115. Thus as the ambient noise levels increase, the gain of the
system for this noise can also increase, resulting in less overall noise
reduction.
Accordingly, it is desirable to provide a method and system for signal
expansion with improved noise rejection capability.
SUMMARY OF THE INVENTION
The present invention provides a desirable method and system for
discriminating between desired sounds from a near-field acoustic source
and sounds (noise) from far-field acoustic sources. The invention
advantageously prevents gain from being added to the undesired (far-field)
loud noises while allowing gain to be added to low-level sounds from the
desired (nearby) acoustic source. As a result, the invention can reduce
the "noise pumping" problem that is encountered by conventional systems
and methods.
The present invention can be used to enhance the noise rejection capability
of headsets. Additionally, the invention may be used in other
applications, such as handsets, as long as two microphones can be placed
at different distances from an acoustic source. The invention may also be
used to enhance the noise rejection capability of voice recognition,
public address, and recording systems.
In one embodiment of the present invention, the signal expander system
comprises an input block, a proximity estimation block (e.g., a ratio
detector) coupled to the input block and a variable gain block coupled to
the proximity estimation block. A speech activity detector may be
optionally coupled to the proximity estimation block.
The input block has at least two inputs (such as two microphones) and three
outputs (such a signal output and the individual outputs of each
microphone). In one embodiment, the signal output of the input block is
the same as the output of the first microphone. In another embodiment,
each microphone output is coupled to a corresponding sensitivity
adjustment block for closely matching the sensitivities of the
microphones. Thus, the outputs of the microphones are derived from their
corresponding sensitivity adjustment block. The signal output of the input
block is derived from the difference between the outputs of the first
microphone and the second microphone. This second embodiment results in a
composite directional microphone from the microphone pair in the input
block. This embodiment is essentially independent of the proximity issue
between the microphones and the acoustic source, and is a convenient
by-product of the above two microphone topology. The delay in the output
signal from the second microphone alters the polar pattern from
bi-directional to cardioid, supercardioid, or hypercardioid, depending on
the amount of delay.
The proximity estimation block includes two inputs (from the microphones)
and one output for generating the proximity estimation block output
signal. As described below, differing degrees of signal conditioning may
be applied in series with the input and output paths of the proximity
estimation block. The proximity estimation (i.e., the distance between
each microphone in the input block and the acoustic source) is made, for
example, based on the ratio between the two inputs of the proximity
estimation block. However, other methods may also be used to make the
proximity estimation.
In one embodiment of the proximity estimation block, the first input
permits the output signal of the first microphone to be received by one
input of the divider, while the second input permits the output signal of
the second microphone to be received by the other input of the divider.
The output signal of the proximity estimation block is derived from the
divider output signal.
Two types of signal conditioning blocks may be interposed on the input side
of the divider. For example, filters may coupled to the divider input side
to band-limit the signals received by the divider. Rectifying detectors
may also be coupled to the divider input side to simplify the operation of
the divider.
In one implementation, the divider is configured to take the difference of
the logarithm of the divider input signals. This implementation is
particularly suited to an analog implementation. The output of this
implementation is the logarithm of the ratio. In some applications, the
antilog of this ratio value may be taken. However, direct use of the log
output frequently will result in better noise margins and permits use of
simpler circuitry.
On the output side of the divider, a clipping and smoothing function may be
inserted to reduce the effects of phase induced transients. This function
may be implemented in either analog or digital form.
The variable gain block includes two inputs and one output and may be
implemented in three basic forms, as described below. The first input of
the variable gain block is the output signal from the input block. The
second input of the variable gain block is from either the direct output
of the proximity estimation block or from a speech activity detector
having an output derived from the proximity estimation. The output of the
variable gain block is the system audio output. All implementations of the
variable gain block typically includes a variable gain amplifier or an
attenuator, either of which is controlled directly or indirectly by the
inputs to the proximity estimation block and may include components for
controlling the timing of the gain changes. All implementations of the
variable gain block may include a delay element on the block input to
compensate for any delays introduced by the proximity estimation process.
One implementation of the variable gain block includes a conventional
amplitude based expander where the gain of the variable gain block is
determined by an input level when the proximity estimation indicates that
the distance to the acoustic source is within a proscribed distance. Thus,
the gain applied to the output of the first microphone is determined by
both the level of the output signal itself and by the estimate of
proximity (as determined by the proximity estimation block) exceeding a
threshold value. The proximity estimate input to this implementation is
binary regardless of the gain versus input level characteristics of the
expander when activated.
A second implementation of the variable gain block includes a form of
bi-stable expander that assumes a high or low gain state depending on the
binary value of the proximity based speech activity detector. Thus, the
output signal of the first microphone is amplified in response to the
proximity estimate exceeding a threshold value.
A third implementation of the variable gain block includes a variable gain
element providing a gain that is a function of the direct output of the
proximity estimation block. The function may be: (i) linear, (ii)
non-linear, (iii) logarithmic, or (iv) arbitrary. Thus, the gain provided
to the output signal of the first microphone is a function of the direct
output of the proximity estimation block. The second input of the variable
gain block in this implementation is directly coupled to the output of the
proximity estimation block.
The speech activity detector includes one input and one output. The
detector input is connected to the output of the proximity estimation
block and permits transmission of the proximity estimation block output
signal to one input or a comparator. The other input of the comparator is
a proximity reference value that establishes the limit of the distance
within which the desired acoustic source is expected to be located. All
acoustic sources more distant than this value will be considered as noise.
The output of the comparator is the output of the speech activity detector
and provides a binary indication of whether the desired acoustic source is
active.
All embodiments of the present invention typically incorporate all four
blocks (input block, proximity estimation block, variable gain block, and
speech activity detector) in one of two general configurations depending
on whether the variable gain block accepts the proximity estimate directly
or uses the speech activity detector output. Any of the variations of the
input block or the proximity estimation block can be optionally used for
any embodiment disclosed herein. The speech activity detector is typically
the same form in all embodiments. The variable gain block can be any of
the three implementations mentioned above.
Additionally, the present invention provides a signal expander using
digital processing to implement the functions mentioned above. In this
embodiment of the invention, the output signals from the microphones in
the input block are first digitized. Analog processes are replaced by
digital arithmetic blocks and algorithms. This digital implementation also
enables more sophisticated processes to be utilized without requiring
additional hardware.
BRIEF DESCRIPTION OF THE DRAWINGS
FIG. 1A is a functional block diagram of a conventional bi-stable signal
expander system.
FIG. 1B is a functional block diagram of a conventional variable gain
signal expander system.
FIG. 2A is a functional block diagram of a signal expander system in
accordance with an embodiment of the present invention.
FIG. 2B is an illustration of a telephone headset that can implement the
present invention.
FIG. 2C is an illustration of a telephone handset that can implement the
present invention.
FIG. 3 is a functional block diagram of an analog divider in accordance
with the present invention.
FIG. 4A illustrates the simulated source waveform when the source is close
to the first microphone.
FIG. 4B illustrates the output waveform of each microphone when the source
is close to the first microphone.
FIG. 4C illustrates the detected output waveform of the signal from each
microphone and the linear ratio, when the source is close to the first
microphone.
FIG. 4D illustrates waveforms of the logarithm of the signal from each
microphone and the difference between the logarithms, when the source is
close to the first microphone.
FIG. 4E illustrates waveforms of the clipped and smooth logarithmic output
which in some embodiments would represent the output of the proximity
estimation block, when the source is close to the first microphone.
FIG. 4F illustrates the output waveform of each microphone when the source
is distant from the first microphone.
FIG. 4G illustrates the detected output waveform of the signal from each
microphone and the linear ratio, when the source is distant from the first
microphone.
FIG. 4H illustrates waveforms of the logarithm of the signal from each
microphone and the difference between the logarithms, when the source is
distant from the first microphone.
FIG. 4I illustrates waveforms of the clipped and smooth logarithmic output
which in some embodiments would represent the output of the proximity
estimation block.
FIG. 5 is a functional block diagram of a signal expander system in
accordance with another embodiment of the present invention.
FIG. 6 is a functional block diagram of a signal expander system in
accordance with another embodiment of the present invention.
FIG. 7 is a functional block diagram of a signal expander system in
accordance with another embodiment of the present invention.
DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS
Those of ordinary skill in the art will realize that the following
description of the preferred embodiments is illustrative only and not in
any way limiting. Other embodiments of the invention will readily suggest
themselves to those skilled in the art.
FIG. 2A illustrates a functional block diagram of a signal expander system
300 in accordance with a first embodiment of the present invention. The
system 300 is advantageously not activated by far-field noises and allows
the use of a lower threshold within the expander control section 320 to
better pick up low level speech. The signal expander system 300 is capable
of detecting the signals generated by an acoustic source 305 and includes
a first microphone 310 coupled to a variable gain block (stage) 315 and to
an expander control stage 320. The expander control stage 320 is also
coupled to the variable gain block 315. The expander control stage 320 may
be implemented by, for example, the conventional expander control stage
205 used in the conventional variable gain signal expander system | | |
