A New Signal Processing Technique for the Estimation of DPOAE Signals
Alireza K. Ziarani
Department of Electrical
and Computer Engineering, Clarkson University, Potsdam, NY, USA 13699-5720
E-mail: Alireza K. Ziarani
Telephone: (315) 268-4278
Fax: (315) 268-7600
1. Introduction
Distortion product otoacoustic emissions (DPOAEs) are very low level
stimulated acoustic responses to two pure tones presented to the ear canal.
DPOAE measurement provides an objective non-invasive measure of peripheral
auditory function and is used for hearing assessment especially in newborns
[1]. DPOAEs have been recognized for a number years [2, 3]. However, DPOAE
measurement is considered an active area of research because of the challenging
nature of the signal processing task.
In this type of otoacoustic test, two pure tones with frequencies
f1 and f2 are presented to the cochlea. For best results,
f2 is usually chosen as 1.2f1. Since the ear is a nonlinear
structure, a number of very low level distortion products are generated due
to the inter-modulation process within the cochlea. Among various distortion
products, the component with frequency fd=2f1-f2
is usually the strongest. The level of such a distortion product (commonly
referred to as the DPOAE signal) is taken as an index of the functionality
of the ear. Estimation of such a weak signal buried under two strong stimuli
and other inter-modulations in a potentially noisy background is a challenging
signal processing problem.
Conventionally, the discrete Fourier transform (DFT) has been used
as the main signal processing tool to estimate the level of the DPOAE signals.
Application of the DFT to this problem has a number of shortcomings, among
which the long measurement time is the most pronounced one [4]. Long measurement
time is usually required for the acquisition of a sufficiently large amount
of data which, when averaged, will reduce the overall background noise effect.
In addition to the need to increase the measurement time, the tests are usually
required to be conducted in low noise environments such as sound-proof rooms
or other types of sound-proof enclosures.
In an attempt to devise high performance DPOAE estimation techniques,
adaptive signal processing techniques and maximum-likelihood estimators have
been employed. Such techniques generally offer better performance in terms
of measurement time which may be interpreted as their higher noise immunity
compared to that of the DFT.
This paper presents an overview of a recently developed method of
DPOAE signal measurement, which employs, as its main building block, a recently
introduced nonlinear adaptive signal processing algorithm [5]. The formulation,
mathematical properties and DSP implementation of the employed signal processing
algorithm are presented in [6] where detailed discussions on the stability
and convergence issues of the algorithm are also presented. Some of the applications
of the employed algorithm in diverse areas of engineering are presented in
[7, 8].
2. Method
The proposed signal processing method employs three units of the algorithm
presented in [6] to construct a high performance DPOAE estimation algorithm.
Each unit is capable of focusing on and extracting a pre-specified sinusoidal
component of its input signal which may contain other components and noise.
More importantly, it can effectively follow variations in the amplitude, phase
(and frequency) of the extracted sinusoidal component. Although the underlying
mathematics ensuring the stability and performance of such an algorithm is
very complex, its structure remains extremely simple. It is found to be very
robust with respect to variations in the internal settings of the controlling
parameters, as well as external conditions such as the presence of noise,
and exhibits superior performance over existing linear adaptive and DFT-based
algorithms in terms of convergence speed versus accuracy trade-off.
The input DPOAE signal is often assumed to consist of two pure sinusoids
with frequencies f1 and f2 at a very high level (usually
about 50 to 70 dB SPL) and a very low level DPOAE 2f1-f2
at about -5 to 15 dB SPL. It is contaminated by a noise usually considered
to be about 0 to 20 dB SPL. In fact, the noise represents the totality of
all undesired signals that may be present in the environment in which the
examination is being conducted, the sum of all generated inter-modulations
as well as the unavoidable background noise. It has been observed that the
estimation error increases with the increase of the amount of background noise.
This can be compensated by re-adjusting the parameters of the algorithm to
reduce the error at the expense of the convergence speed. Because of the excessive
degree of the noise, one single unit assigned to extract the DPOAE signal
out of the input signal exhibits poor performance in terms of the estimation
error, (or the convergence speed).
Figure 1 . Block diagram representation of the proposed method of DPOAE
estimation.
In the block diagram of Figure 1, the first two units are assigned
to extract the two stimuli. They effectively do so with very small errors.
The extracted stimuli are then subtracted from the input signal to produce
a signal, of which DPOAE has a higher relative portion. The third unit is
then set to extract the DPOAE signal. To further enhance the performance of
the DPOAE estimator, some pre-processing, post-processing as well as intermediary
filtering stages have been added. The stage of pre-processing consists of
preliminary normalization and band-pass filtering. The purpose of the normalization
process is to amplify the input signal to bring it to a certain level on the
basis of which the setting of the parameters of the units may be adjusted.
The band-pass filtering is intended to attenuate all components except the
DPOAE signal as much as possible to enhance the quality of the input signal.
This can be achieved by means of a simple second order band-pass filter, the
center frequency of which is set to be that of the DPOAE signal.
The intermediate signal out of which the two stimuli are removed may
be directly input to a third unit for the extraction of the DPOAE signal.
Since elimination of the two stimuli needs certain convergence time, at the
very early initial moments a large portion of the two stimuli exists which
will set the initial operational point of the third unit too far away from
the true level of the DPOAE signal. To overcome this, a time-gating process
may be employed to delay the transfer of the intermediate signal to the third
unit. This is accommodated in the mid-processing unit of Figure 1. The output
of this unit is zero and remains zero for a short period of time until a more
or less steady state condition for the two units is achieved. The mid-processing
may also include some normalization and band-pass filtering just like the
pre-processing stage. The post-processing unit consists of de-normalization
of the DPOAE signal and its level to restore the original values as well as
some (low-pass) filtering to further smooth out the estimate of the DPOAE
signal and its level.
3. Results
Performance of the proposed method is demonstrated in this section
using both flexibly controlled simulated data and a set of real clinically
recorded signal.
3.1 Simulated Data
For the simulations presented in this section, the frequency of the
first stimulus f1 varies over the range of 500 to 4000 Hz. For
each numerical experiment, f1 is randomly generated within this
range. The frequencies of the second stimulus and the DPOAE are then set as
f2=1.2f1 and fd=2f1-f2,
respectively. The initial phases of the simulated stimuli and the DPOAE are
randomly chosen within 0 to 180 degrees. The simulated noise added to the
input signal is a zero-mean white Gaussian noise for the first two experiments
and is a pink Gaussian noise for the third experiment. The levels of the stimuli,
DPOAE and noise floor are specified in each case.
Figure 2 presents the performance of the proposed DPOAE estimator
when the levels of the two stimuli are randomly generated within the range
of 0.8 to 1 V (roughly corresponding to a relative 60 dB level) while the
DPOAE signal has a level of about 6 mV (corresponding to a relative 15 dB
level). The noise floor is at about 10 dB. The conditions in this example
are in accordance with the defined conditions of the problem. It is observed
that the convergence is achieved well within the desired one second test period
with a small estimation error. In another numerical experiment, the level
of the noise floor is increased about four times (corresponding to about 28
dB level). Figure 3 shows the estimation process. The estimation is achieved
well within the desired one second test period with a tolerable estimation
error. The present parameter setting easily accommodates noise levels of up
to 30 dB, which is believed to be an exaggeration of the actual scenarios.
However, if the expected noise floor happens to be even higher, one can sacrifice
the speed by re-adjusting the parameter settings. Generally, one needs to
take into account the following factors when choosing the values of parameter
settings: some idea about the potential background noise, the desired speed
of convergence and the tolerable error. Experience of the authors as well
as that of the collaborators in companies manufacturing DPOAE measurement
equipment confirms the suitability of the suggested parameter settings in
practical DPOAE measurement tests.
Figure 2 . Illustration of the performance of the proposed DPOAE estimator
using simulated data.
Figure 3 . Illustration of the performance of the proposed DPOAE estimator
using simulated data. Compared to the conditions of Figure 2, the noise floor
in increased about four times.
Typical environmental noise experienced during the recording of signals
within the acoustic range is pink, in which the power density decreases with
increasing frequency over a finite frequency range so that each octave contains
the same amount of power. To present a more realistic demonstration of the
performance of the proposed method, the experiment of Figure 3 is repeated
using a pink background noise. Figure 4 illustrates the estimation process
where f1 is about 3500 Hz. It is observed that the performance
of the proposed method is better in this case as opposed to the case of the
contamination by white noise (Figure 3). This can be explained by the fact
that given the non-uniform distribution of the noise power, less noise exists
around the frequencies of interest. This renders the estimation process more
accurate. Further experiments show that at lower frequencies, the effect of
pink noise is more destructive than that of the white noise, as would be expected
theoretically.
Figure 4 . Illustration of the performance of the proposed DPOAE estimator
using simulated data. This is the same experiment as that of Figure 3, but
with pink background noise.
Figure 5 . Illustration of the performance of the proposed DPOAE estimator
using a set of clinically recorded data.
3.2 Recorded Data
One set of clinical data recorded at Rotman Research Institute of
the University of Toronto is used to verify the functionality of the proposed
method. The recording is conducted using specialized otoacoustic probes. About
20 seconds of recording is available. The total length of the recording is
used to obtain the frequency spectrum of the signal, which in turn can serve
as a means of guessing the true value of the DPOAE level. The frequencies
of the two stimuli and the DPOAE are f1=1618, f2=1797
and fd=1438 Hz, respectively. Figure 5 presents the performance
of the proposed method. It is observed that the convergence is achieved within
the desired one second test period with a small estimation error.
Figure 6 . Comparison of the performance of the proposed method with
the method of Ma and Zhang [9].
4. Discussion
One of the recently proposed methods presented by Ma and Zhang in
[9] is used in this section for a quantitative comparison with the proposed
method. The method presented in [9] is an optimal maximum-likelihood estimator
for the extraction of DPOAE signals. Superior performance of the method, especially
in cases where DFT exhibits leakage effect is observed. The signal model is
assumed to consist of the two stimuli and the DPOAE signal with noise. In
[9], simulated data are used for the two stimuli and the DPOAE signal whereas
the background noise is a recorded noise.
Simulated data were used for the comparison of the method of Ma and
Zhang with the proposed method. Similar results to those presented in [9]
were obtained using a simulated noise of zero-mean white Gaussian distribution
as the background noise. For both cases, the numerical experiments involve
two stimuli of frequencies f1=2.454 kHz and f2=3.003
kHz. The DPOAE signal is thus at fd=2f1-f2=1.905
kHz. The sampling frequency is chosen as fs=10.24 kHz. The DPOAE
signal is at 0 dB level while the two stimuli are at 65 dB. The experiment
was repeated several times for different levels of the noise floor. Figure
6 compares the performance of the two methods for varying levels of the noise
floor. The index of the performance is taken to be the normalized mean squares
error (MSE) incurred in the estimation process. In the case of the proposed
method, the vector of the estimated level of the DPOAE is formed after the
signal is stabilized in the time domain (after about 500 ms delay).
When the incurred error exceeds the signal level, the signal is no
longer recoverable. It is observed that the proposed method is less sensitive
to the level of the background noise. In fact, as soon as the noise level
exceeds the DPOAE level, the signal is totally lost and the estimation process
fails. The proposed method of this paper has a very high degree of noise immunity,
about 20 dB more than that of Ma and Zhang.
5. Conclusions
A method of measurement of DPOAE signal level employing a recently
introduced nonlinear adaptive signal processing technique is presented. Performance
of the proposed method is demonstrated using both simulated and real clinical
data, and a comparison of its performance with that of one of the existing
methods is presented. The main features of the proposed method of DPOAE signal
measurement are its 1) structural simplicity, 2) high noise immunity and robustness,
and 3) relatively high speed of convergence. Given the low complexity of the
proposed method, it requires low level of computational resources, which in
turn translates into less expensive equipment. High noise immunity and robustness
of the proposed method render it suitable for practical clinical examinations
which may be conducted in highly noisy backgrounds, perhaps without involving
sound-proof examination rooms. This again translates into less expensive medical
equipment. Also, given that the reduction in the level of the stimuli translates
into a higher relative degree of background noise, the high noise immunity
feature of the proposed method may be used to reduce the level of the stimuli
for a more patient friendly examination. High speed of convergence of the
proposed method is useful in reducing the examination time which again results
in a more patient friendly and time effective clinical examination.
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