Abstract The main complaint of hearing instrument wearers continues to be hearing in noise. While directional microphone technology exploits spatial separation of signal from noise to improve listening in noise, a challenge for single microphone noise reduction systems is that the signal of interest is embedded in the background noise. ReSound NoiseTracker™ II uses a sophisticated algorithm to reduce amplification of noise for both single and dual microphone devices without impacting speech understanding. It accurately identifies speech and characterizes noise, operating seamlessly to reduce noise even during the pauses in running speech. In the clinic, ReSound NoiseTracker™ II can be customized for different listening preferences and/or listening situations.
Unwanted amplification of background noise is a
usage and satisfaction with amplification. Nabelek
significant factor influencing hearing instrument
and her colleagues proposed that hearing
user satisfaction (Kochkin, 2000). Just as
instrument usage might be determined more by
directional microphone technology has risen
wearers’ willingness to listen in background noise
enormously in popularity as a means to combat
than how well they understand speech (Nabelek et
hearing-in-noise issues, noise reduction algorithms
al, 1991). Using the acceptable noise level (ANL)
have become an ever-present feature in digital
test with elderly listeners, they found that full-time
background noise than did part-time users or users
This paper reviews the standard methods of single
who had rejected their hearing instruments
microphone noise reduction and describes the
(Nabelek et al, 2006). These findings would
ReSound NoiseTracker II system, an important
suggest that eliminating or reducing background
component of the surround sound by ReSound.
noise when listening with hearing instruments
NoiseTracker II is unique in its ability to reduce
would increase the acceptance rate of hearing
amplification of noise without affecting the
instruments and provide significant benefit to a
audibility or quality of simultaneously occurring
speech. This provides not only increased listening
Perhaps related to the willingness to listen in noise,
listening effort in background noise also has far-
Even though hearing instrument users of all ages
instruments. Consider, for example, the hearing
complain about background noise, it can be an
instrument wearer who is attempting to listen to
especially problematic issue for older adults, who
speech in a noisy environment over a prolonged
comprise the largest segment of users. The
period of time. The concentration required to
average hearing instrument user is 70 years old
follow what is being said can be an exhausting
(Kochkin 2005). In addition to hearing loss, many
task. Conceivably, less effort would be required by
older adults often experience problems with
the listener if the background were less noisy, even
auditory processing and cognitive function, which
if overall understanding of speech was not
can result in difficulty focusing, remembering or
improved. Less effort required of the hearing
processing information they have heard (Wingfield
instrument user may even allow them to perform
2005). Studies have shown that even when younger
dual attention tasks that normal listeners typically
and older adults have similar hearing thresholds
take for granted, such as listening to a talker while
and speech discrimination ability in quiet, older
also paying attention to their surroundings,
adults will perform worse on speech discrimination
simultaneously noting other conversations that may
tasks in noise than their younger counterparts
be going on around them, or monitoring the activity
(Kricos 2006). In addition, Tun et al (2002)
of other people in the room. Simply eliminating a
suggested that higher level cognitive factors may
distracting or annoying sound, such as a nearby air
affect the ability of older adults to process language
conditioner could, in other words, provide
in the presence of competing speech. In terms of
significant benefit to a hearing instrument user
hearing instrument use, understanding speech in
noise is likely to be particularly challenging for the
older adult, even if audibility is provided. This in
turn impacts success with amplification.
Noise reduction and dual microphone directionality
in hearing instruments were both conceived as a
Apart from issues related to processing speech in
means to improve speech understanding in noise.
noise, other noise-related factors can influence
Directionality achieves this goal by preferentially
amplifying signals coming from a particular
were based on signal modulation and are in fact
location, such as in the look direction of the
still the most common type of system today.
hearing instrument user. For noise reduction to be
Modulation-based noise reduction systems analyze
effective in improving speech understanding in
the level fluctuations of the input signal in different
noise, it would have to separate desired speech
frequency bands. Because single-talker speech can
from the competing signals after both have been
fluctuate in level more than 30dB, the assumption
picked up by the same microphone. Even then, the
is made that large fluctuations in the input sound
hearing instrument would need to be capable of
indicate good signal-to-noise (SNR) ratios, and that
reducing amplification of only the noise
components of the incoming sound without
fluctuations decrease. If the SNR is estimated to be
disturbing the speech components. This is a
poor in a particular frequency band, gain is reduced
daunting task, as both speech and what constitutes
“noise” for the listener are similar in terms of
frequency content. In fact, the undesired noise
Although the underlying principle is the same,
often is speech from multiple talkers. Not
modulation-based systems differ in a number of
surprisingly, the benefit of noise reduction has been
important ways. One obvious difference is the
noted more in its ability to improve listening
amount of gain reduction. Some systems may allow
comfort and effort than for improving speech
more than 10dB reduction in gain while the
understanding (Mueller et al, 2006; Bentler et al,
maximum reduction for others can be 5dB or less.
Another difference is that some algorithms may
consider other acoustic properties of the input than
Prior to digital hearing instruments, noise reduction
just modulation in determining the amount of gain
systems limited low frequency amplification by
reduction. For example, they may reduce the gain
means of a high pass filter or by means of different
differentially depending on the rate of modulation
compression strategies. The rationale behind this
or on the overall input levels. Finally, systems
approach was that noise – including multitalker
differ in how long they take to decrease gain or
babble - contains more energy at low frequencies
restore it back to its original value in a given
than the speech of a single talker, which could be
channel. As in describing the dynamics of
expected to contribute significantly to excessive
compression systems, these are collectively called
overall loudness of the signal. Also, low frequency
the time constants. Systems which reduce and
noise may cause upward spread of masking,
restore the gain slowly can make listening in
thereby making the high frequency parts of speech
situations with stationary noise sources more
comfortable, but may degrade the audibilility of
speech. For this reason, most systems incorporate
The advent of digital technology enabled more
faster release times so that the gain is restored
advanced means of determining the composition of
quickly when a highly modulated signal such as
the input signal. A better characterization of the
speech is detected. Systems which also have fast
sound entering the hearing instrument microphone
attack times may negatively affect the overall
could make it possible to limit gain reduction to
sound quality as gain is rapidly increased and
periods of time and spectral regions where
important speech information was occurring. The
first digital noise reduction algorithms to appear Figure 1. Steady-state noise is treated differently by different noise reduction systems. Figure 2. For babble noise, modulation-based noise reduction systems do not affect the background noise. NoiseTracker™ II is different than traditional modulation-based systems as it is able to reduce the level of the babble while preserving the level of speech. As a result of such differences, modulation-based
applications. The concept of spectral subtraction,
noise reduction systems also differ greatly in how
illustrated in Figure 3, is to subtract the short-term
various types of input sounds are treated (Bentler &
noise spectrum from the total signal, leaving only
Chiou, 2006). An example of this is illustrated in
the speech portion. Although the concept is easy,
Figure 1 which shows how different types of
the implementation is not. The success of this
system react to steady-state noise with a short
strategy hinges on being able to identify speech and
speech passage at the end of the timeframe. Figure
to precisely characterize noise. An additional
challenge is to keep up with the dynamic speech
background noise and the same short speech
and noise make-up of real listening environments.
passage at the end of the timeframe. There are
Finally, it is important for hearing instrument users
clear differences in both reaction time as well as
that not all noise be removed from the signal, and
amount of gain reduction for the same input. While
that the noise characteristics be preserved. If all
no evidence exists supporting the superiority of any
ambient noise were removed or if the spectrum of
particular system, there is one apparent
the noise background was altered, this would create
shortcoming of the modulation-based approach.
an unnatural-sounding experience. Background
This is the inability of such a system to accurately
sounds do need to be audible to the degree that
identify when speech is present. As a result,
users can recognize and orient themselves in their
modulation-based systems are more likely to make
listening environments. Ultimately, the goal is
“mistakes” in terms of when and in what
undistorted speech at the prescribed gain, and
frequencies gain reduction would be beneficial.
Like other noise reduction schemes, NoiseTracker
II has the goal of suppressing noise in frequency
regions where the speech-to-noise ratio is low.
However, NoiseTracker II is distinguished from the
modulation-based method by its ability to reduce
unwanted noise from the incoming signal without
appreciably affecting the speech portion of the
signal. The NoiseTracker II system is able to
accomplish this because of 1) the higher degree of
accuracy with which it identifies speech and noise
when compared to other systems, 2) an adaptive
Figure 3. Spectral subtraction removes noise from
noise estimate, 3) fast time constants, and 4) a
the total signal, leaving the desired signal intact.
mathematically optimal gain reduction function
based on SNR rather than signal modulation.
The NoiseTracker II signal analyzer itself is
comprised of three main components: a signal
Built on the ReSound Warp-based platform, the
power tracker, a speech presence indicator, and a
NoiseTracker II system uses spectral subtraction
noise power tracker. These three components
(Boll, 1979), one of the most widely used methods
provide information that is used to estimate the
for enhancement of noisy speech in audio
signal-to-noise ratio (SNR), which in turn
determines the amount of gain reduction to be
employs time constants for each of its tracking
components as well as for the actual gain reduction
and restoration. While the signal power tracker
The signal power tracker represents the overall
always works quickly in order to preserve the
signal including speech and noise, and is the part
speech envelope, the noise power tracker
from which any noise will ultimately be subtracted.
adaptively adjusts its time constants depending on
The speech presence indicator analyzes the acoustic
whether speech is detected. Estimation of the noise
characteristics of the signal at 1-millisecond
spectrum is thus limited to pauses between words
intervals to determine the probability that speech is
and syllables. In this way, the system avoids
present in the overall signal. More specifically, the
mistaking speech for noise, and prevents speech
speech presence indicator looks for a temporal-
information being subtracted from the overall
spectral pattern of alternating high and low
input. Once noise is detected, the time it actually
frequency sound which is typical of speech. This
takes for the decrease in gain to begin is within
method constitutes a more precise way of
2 seconds. As the SNR improves or decreases
identifying speech than relying on modulation
further, new gain calculations are effected almost
alone. With such an accurate identification of
speech, the noise power tracker is able to restrict
analysis of the noise background to frequency
regions and points in time where speech is not
NoiseTracker II system offers flexibility in the
mixed with noise. It is critical for the system that
degree of noise reduction it offers to address
the noise estimate not be contaminated by speech,
individual user preferences. Up to four options are
since this would lead to distortion of the speech
available: mild (-3dB), moderate (-6dB),
considerable (-8dB) or strong (-10dB) noise
reduction. The noise reduction value for each
Once the overall signal and noise information have
degree of NoiseTracker II is the amount which
been provided by the speech presence indicator and
would be applied when the estimated SNR is 0dB
the noise power tracker, the SNR ratio is estimated
or worse. A lesser amount of gain reduction would
by comparing the level of the noise with the level
be applied for better SNRs as illustrated in
of the overall signal. When only noise is present in
the total input signal the difference between the
NoiseTracker II setting while others prefer being
total input signal and the estimated noise will be
able to hear more of the environmental noise.
small, as will the SNR estimate. Conversely, the
Preferences regarding degree of noise reduction
SNR estimate will be greater when the total input
may also differ from situation to situation, which is
signal consists of both speech and noise.
the rationale for the different NoiseTracker
Depending on the estimated SNR and the user-
settings in many of the environmental programs.
configurable NoiseTracker II level setting in the
For example, the “Traffic” program is set to apply
Aventa software, the gain may be reduced as
a strong level of noise reduction, as it is assumed
illustrated in Figure 4. The gain reduction functions
that maximum listening comfort would be desired
are mathematically derived using Wiener optimal
in this situation. All degrees of NoiseTracker II can
filter theory. When noise and speech are present
be applied without affecting speech intelligibility
simultaneously, the most recent noise estimate is
limitations of modulation-based noise reduction
systems. Because of the accuracy with which is
identifies speech and noise, it excels in its ability to
reduce unwanted noise from the incoming signal
without affecting audibility of speech or sound
quality. With NoiseTracker II, the ReSound
hearing instrument wearer can hear desired sounds
while noise is kept at comfortable levels to allow a
Figure 4. The NoiseTracker gain reduction function is based on the signal-to-noise ratio
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The time constants of the NoiseTracker II system
are crucial for its performance. This system
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APPLIED PHARMACOLOGY COURSE TITLE: MW 6420 APPLIED PHARMACOLOGY Course hours: 25 hours (5 classes) each class is 5 hours Course Description: This course will cover the use, preparation and effects of drugs relative to pregnancy, labor, birth, postpartum and the neonate Course Objectives: Given pertinent information, supported by instructional resources, the student will be capabl