Ala.
3
A GENERAL-PURPOSE HEARING AID PRESCRIPTION, SIMULATION AND TESTING SYSTEM
Donald G. Jamieson and Emmet Raftery
Speech Communication Laboratory
Department of Communicative Disorders, University of Western Ontario
ABSTRACT
This paper describes a comprehensive, integrated,
microcomputer-based facility for research on hearing aid
fitting, and for clinical use. The system first applies
formal prescription rules to audiometric measures which
have been obtained from an individual patient to predict
which hearing aid amplification function is likely to
provide the optimal benefit for a hearing impaired
listener. It then designs and implements a filter in a
digital-signal processing (DSP) board, which precisely
implements the desired gain function. Speech
intelligibility and listener preferences can then be
measured to evaluate the prescribed hearing aid gain
function using efficient, computer-based testing
procedures. The system requires an lBM/AT compatible
microcomputer (eg., Zenith Model 386 or 248, with math
coprocessor and EGA or VGA graphics), an ARIEL DSP--16
digital signal processing board, plus filters and
associated analog audio equipment, and a Mouse.
The precise relation between a hearing-impaired
listener's performance on basic audiometric tests and
the amplification characteristics which would lead to
the highest level of speech understanding for that
listener remains an open question. A wide variety of
procedures have been proposed to guide hearing aid
prescription [ I ,2,3,4,5,6,7]. These methods differ in
both the form and manner in which audiometric data are
obtained, the operational definition of the spectra and
levels of conversational speech and the computational
rules used to predict the "optimal" amplification
characteristics for a give listener.
To evaluate the alternative prescription
procedures, four conditions must hold. First, since a
variety of audiometric testing procedures are used in
the various prescription procedures, these should be
evaluated with respect to their ability to provide data
which are valid, reproducible, and predictive of an
individual listener's ability to understand speech.
Second, different prescription rules make different
predictions concerning such factors as the spectrum of
the speech stimuli which the listener will hear, the
coupler-to-ear canal transfer function, and the field-
to-ear canal transfer function. These assumptions
should be evaluated within the particular testing
situation which is used to evaluate the rules. Third,
the amplification functions which are predicted to be
optimal for each listener must be implemented precisely,
and the experimenter must confirm that the planned
amplification gain function was in fact the one
implemented during testing. Fourth, the criterion used
to identify different levels of speech understanding for
the listener must be valid and reliable, if one is to
identify which of a set of such amplification functions
is "best" according to a well-defined criterion.
1989
In all previous attempts to evaluate alternative
prescription rules, one or more of these conditions has
been violated. The present paper describes a system
which we have developed in order to evaluate alternative
hearing aid prescription rules, and ultimately, to
develop and refine more optimal rules.
Distinctions Between Alternative Hearing Aid
PrescriDtion Rules
The first way in which hearing aid prescription
rule systems differ is in the audiometric data which
they require. Some require only threshold measures,
while others require one or more estimates of the
listener's most comfortable listening levels (MCL)
and/or an estimate of the listener's uncomfortable
listening level (ULL). Even when two rules do require
the same type of measure, the manner in which these data
are required to be collected may differ. A summary of
the audiometric measures required by the various rules,
and a description of the testing conditions, in each
case, is provided in Jamieson, Seewald and Raftery [SI.
The second way in which prescription rules differ
is in their specification of the spectrum of the speech
stimuli which the listener will hear. These differences
occur not just in the particular samples of speech used
by different investigators. but also in the way in which
the spectrum was measured. This factor introduces
substantial variation across prescription methods.
Finally, what is most distinctive about alternative
hearing aid prescription rules is the particular way in
which the rules combine the data obtained from the
patient with the measurements associated with the
listening condition. Threshold-based prescription
strategies have been developed by Berger [I], McCandless
and Lyregaard 191. Byrne and Dillon [2] and Seewald and
Ross [ 5 ] . Two approaches which require direct (supra-
threshold) measurement of comfortable loudness, in
addition to threshold, have been described by Cox [4,
IO] and Skinner, et al. [7]. Each of these various
approaches will prescribe different frequency/gain
characteristics for the same audiometric profile (i.e.,
for the same listener, cf., [ I 1,121).
Of the four threshold-based approaches, three
employ some variation of the "half-gain rule" to develop
the prescription. Of current approaches, the
prescription of gainjoutput (POGO) method described by
McCandless and Lyregaard [9] closely approximates the
half-gain rule, by multiplying the hearing level at all
audiometric frequencies by 0.5 and subtracting IO and 5
dB from the obtained values at 250 and 500 Hz,
respectively. McCandless and Lyregaard have reportedly
included the reduction in the prescribed low frequency
gain in an attempt to limit the upward spread of masking
effect from low-frequency ambient noise. Berger [I]
proposed a set of prescription rules in which the
individual's hearing levels at the octave frequencies
250-4000 Hz are multiplied by 0.45, 0.5, 0.625, 0.667,
and 0.5, respectively.
 The procedure developed at the National Acoustic
Laboratories (NAL) in Australia by Byrne and Dillon 121
is the only attempt to date, to account for both degree
of hearing loss and slope or configuration of hearing
loss, in developing the frequency/gain prescription.
The NAL approach uses a more complex linear equation,
of the form TAf = .3I*,Tf + X - Cf. where TAf is the
target amplification (desired real ear gain at
frequency f), Tf is the listener's threshold at
frequency f, measured in dB HL, Cf is a frequency-
dependent constant which varies between -17 dB and + I
dB, and X = O.OS*(Tsoo + T ~ O O +O T2000)/3 -- i.e., .OS
times the three-frequency average threshold, in dB HL.
A somewhat different threshold-based approach was
developed by Seewald and Ross [SI for use with young
children. This approach uses estimates of the desired
sensation levels (DSLs) to which speech should be
amplified; these levels vary as a function of both
hearing level and frequency region. By targeting levels
to which speech should be amplified, the prescribed
frequency/gain function is equal to the amount of gain
which would be needed to amplify the long-term averaged
speech spectrum to these desired levels.
In contrast to the strategies described above,
several prescriptive approaches have been developed
which require the direct measurement of various supra-
threshold loudness dimensions. The C.I.D. Phase IV
method [7] for example, seeks to ensure sufficient
amplification so that normal conversational speech is
placed near the listener's most comfortable listening
level. The specified amplification is given by TAf
(MCLf - Tf)*Pf + Tf - Sf. where TAf is the target
- channels [la]. Compensation for frequency response is
amplification (desired real ear gain, in dB) at
frequency f, MCLf is the arithmetic average most
comfortable listening level at that frequency, Pf is a
multiplier between .5 and I , and Sf. is an estimate of
the level of the speech spectrum at frequency f.
Essentially, the procedure measures the difference, in
dB, between the listener's threshold and most
comfortable listening level (at each octave and
interoctave frequency between 250 Hz and 6,000 Hz) adds
a proportion of this difference to the listener's
threshold, measured in dB HL, and then corrects this
level of amplification by an amount equivalent to an
estimate of the average level of speech within each
frequency band.
mn-
Our present version of the hearing aid
prescription, simulation and testing system was
developed from that described by Jamieson, et al.
[13,14]. Briefly, we have implemented, on an IBM/AT-
compatible microcomputer, an integrated system which
accepts audiometric data for hearing-impaired listeners,
generates a prescription for the hearing aid gain
function specified by the desired prescription rule, and
designs a digital filter which will implement that gain
function on a digital signal processing DSP chip
(TMS32F2S on the ARIEL DSP-16 board)I.
PrescriDtion Rule Svstem
In order to facilitate the process of generating
prescriptions for hearing aid gain functions, we have
developed a comprehensive system for patient database
records, and for computing each step in the process of
fitting a hearing aid [IS]. After performing the
calculations specified by a given prescription rule, the
program specifies the desired, real-ear gain for the
patient.
Piaital Signal Processinn System
The gain function specifications generated by the
Prescription Rule System provide the input for a DSP
filter designjcontrol program. This program uses
measures of the electroacoustic environment (i.e., the
equipment and transducers) and of the stimulus source to
design a digital filter which will realize precisely the
intended prescription (i.e., gain function). The
1990
program then initializes a Real-Time Executive (RTX)on
Ariel's DSP-16 and designs the filters realize the
required frequency response function.
The filters are designed using a frequency-sampling
method to obtain linear phase, Finite Impulse Response
(FIR) filters. The control program then generates the
machine code to implement the filters as interrupt
handlers. The code generation software optimires the
resolution of the coefficients and compensates for the
scaled coefficient values on the filter output. The
optimized code is downloaded and installed in the memory
of the DSP-16. The RTX running on the DSP-16 will allow
several s e a of coefficients and code to be downloaded
from the IBM PC (host). On command from the host, the
RTX running on the DSP-16 installs the appropriate
filter, using the desired set of coefficients. With
this approach, it is possible to switch between filter
responses instantaneously.
The filter design/control program on the host can
monitor the actual signal delivered to the subject's
eardrum via a high-quality probe-tube microphone placed
in the subject's ear canal. Hence, the program can
measure the frequency response of the complete
electroacoustic coupling to the eardrum. To do this, we
output known complex signals from the DSP-16 through the
output transducer, (usually an insert earphone), and
then measure the response at the probe-tube microphone.
The response from the probe-tube microphone is pre-
amplified and calibrated through one of tlie DSP-16's A/D
included in the design of each of the hearing aid
prescription filters.
The probe-tube subsystem is also used to measure,
set and verify the actual gains of the digitally
controlled output amplifiers of the system. Thus, the
actual calibration. verification, and filter design
procedure, which by previous methods was cumbersome,
slow, and often inaccurate or impossible to perform, can
now be performed automatically in seconds. Since the
system can instantaneously toggle between filters, it
can be used to compare alternative hearing aid
prescriptions directly, providing a capability which has
heretofore been unavailable.
Evaluation of Alternative PrescriDtion Rules
To evaluate how well individual listeners
understand speech which is amplified by the different
rule systems, we first obtain complete audiometric data
for individual listeners, in each of the ways required
to apply those rules. We then measure the acoustical
characteristics of our listening situation2. Next, we
combine these data according to the rules specified
under the different systems. Finally, we measure
listeners' abilities to understand speech which has been
processed using the precise gain functions specified by
the alternative rules.
As an example of the differences associated with
collecting audiometric data in different ways, the
auditory thresholds MCLs and LDLs measured for a single
listener, using the different audiometric measurement
procedures specified by the various prescription rules
are presented in Table 1. When these data are combined
in the manner specified by two hearing aid prescription
rules, the required spectrum of speech, measured at the
lvmaaolc membranp, is presented in Figure 1. Thus, the
functions summarize the effects of applying, to our
continuous discourse material, the amplification which
each prescription rule specifies should be provided to a
signal which arrives at the microphone of a behind-the-
ear hearing aid, and is delivered to the listener's ear
canal, and of applying all necessary correctinns for
nonlinearities associated with the electroacoustics and
physics of the listening situation.

Table 1. Audiometric data in dB HL for a single
hearing-impaired listener. Data were obtained using the
recommended measurement procedures specified by four
different prescription methods.
National Acoustic Laboratories (NAL)
FREQUENCY
-
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L
.................................
cr II . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
:................................
FREQUENCY
Berger
Figure I. One-third octave band levels of amplified
speech measured at the tympanic membrane of one hearing-
impaired listener. Data shown are the result of gain
functions prescribed by the N.A.L. (left panel) and
Berger (right panel) methods.
Evaluation of Soeech Intellieibility
Three approaches are involved in our assessments of
listeners' abilities to understand speech which has been
amplified in various ways. The first is a closed-set
format with consonant-vowel (nonsense) syllables, which
evaluates listeners' abilities to identify several
specific speech sounds which are known to cause
difficulty for many hearing impaired listeners [17].
"CL IC!",
The second uses continuous discourse passages to
evaluate listeners' abilities to comprehend continuous
speech [17]. The third is the direct comparison of two
alternative gain functions, achieved by playing speech
through each of the functions in turn, with the listener
able to toggle between the two functions as desired,
before indicating which of the two functions is
suoerior.
~~
Nonsense Svllable Testing
Levitt and Resnick [I81 introduced the Nonsense
Syllable Test (NST) as a w a i to obtain both an estimate
of a listener's relative ability to understand speech
under different listening conditions, and to analyze the
listener's specific difficulties in understanding
speech. The test has since been examined under a
variety of listening conditions (cf., [19,20,21]). For
our version of the NST we recorded, on computer disk, 55
separate consonant-vowel (CV)and vowel-consonant (VC)
syllables3 spoken by a single, male, Canadian talker.
These. signals were then edited, using a general-purpose
waveform editor [22] to ensure that they met our
acoustic criteria, and they were judged by a panel of
normally-hearing listeners, to ensure that they sounded
natural and were readily identifiable.
The 55 syllables are arranged into seven, closed-
set modules, which are presented to listeners
individually. Each test session consists of one
presentation of each of the seven modules; within a
module, syllables are presented' in a randomized order.
with one syllable selected by the program (at random) to
be presented twice, and all other syllables from the
module presented exactly once. Each trial of the test
begins with the presentation, on a computer monitor
screen. of the set of possible responses for the trial.
Next, the target sound is presented to the listener
within the carrier phrase "Point to the sound ...". The
listener then uses the mouse to move a visible pointer
on the monitor screen, to indicate the desired response,
then pressed the mouse button so that the response was
recorded4. The computer then reorganizes the set of
possible responses for presentation on the monitor, and
I-n
.............
begins the next trial. One run through the full NST
requires approximately 8 minutes. At the conclusion of
testing, the computer scores and displays the data.
Jamieson. Dell'Orletta, and Ramji [I71 have presented
data indicating that the NST is very sensitive to
changes in listening condition and that the test results
are highly reproducible.
Continuous Discourse Testing
Tests such as the NST can provide detailed
analytical information concerning the specific types of
difficulties which a hearing-impaired listener may have
with speech. A separate clinical need is to obtain a
rapid measure of a listener's overall level of
understanding of speech, in a particular listening
situation. One attempt to meet this need, involves
direct ratings of the intelligibility of continuous
discourse samples by the listener (following [23]).
Continuous discourse contains different acoustical
cues to the identity of individual words (i.e., dynamic,
coarticulation effects, both spectral and temporal), as
well as lexical. semantic and syntactic cues to word
identity. which cannot be examined in tests using
syllables or isolated words. A variety of alternative
rating approaches have been considered (e.g., [23,24,
25,261). Our approach is to play the listener a 40 to
60 second passage of a story, then have them judge the
percentage of words which they understand correctly.
They respond by using the mouse to move a pointer on
the computer screen, to indicate that percentage on a
continuous scale (from 0 to 100). While such testing
can be undertaken only within a limited range
(performance-intensity functions are considerably
1991

steeper than for the NST, for example), listeners can
make such judgements quickly, and easily, and the
results are relatively orderly and predictive of speech
intelligibility measured as a percentage of syllables
.. . identified
correctly .-. under comparable listening
conditions [XI.
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Cox, R.M. (1988). The MSU hearing instrument
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FOOTNOTES
This work is supported by grants from the Ontario
Premier’s Council Technology Fund and from Unitron
Industries Limited. Earlier work on the project was
supported by Health and Welfare Canada, under the
National Health Research Development Program. We are
indebted to Barb Bentley, Meg Cheesman, J.P. Gagne. Dino
Dell’Orletta, Lynne Powell, Ketan Ramji, Richard
Schmidt, and Richard Seewald for assistance and advice
at various stages of the project.
Correspondence should be addressed to Dr. Donald G.
Jamieson, Speech Communication Laboratory, Department of
Communicative Disorders, University of Western Ontario,
1. Moreover, the system described here is closely
related to the one used to develop CSRE the Canadian
Speech Research Environment. The CSRE software permits
detailed speech recording, editing, analysis, synthesis,
replay, and testing, on an inexpensive, general-purpose
microcomputer-based facility ([22,27]).
2. For example, some prescription rules assume that
speech has a particular power spectrum when the speech
is measured in a specified way, and use this spectrum to
predict the amplification characteristics which would be
optimum for a given listener. If, in evaluating
alternative prescription rules, the listeners are to be
tested with speech having a spectrum which is quite
different from that assumed by the prescription rule.
3. There is evidence that consonant clusters (CCVC,
CVCC, CCCVC) can improve the prediction of listeners’
speech understanding abilities ([XI). Future versions
of our NST will include such clusters.
4. Since this response is under the subject’s
control, the procedure permits the listener to control
the pace of the test.