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JP2010178330

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DESCRIPTION JP2010178330
A method of modeling the feedback path from a receiver of a hearing aid to a microphone, the
provision of a digital feedback suppression circuit that reduces the user's discomfort during the
initialization process. A feedback path based on the steps of transmitting an electronic probe
signal to a receiver to convert it to an acoustic probe signal output by the receiver, recording a
microphone output signal, and the recorded microphone output signal. Determining at least one
parameter of. The steps of transmitting the probe signal to the receiver may include increasing
the level of the probe signal, monitoring the value of the first quality parameter calculated based
on the recorded microphone output signal, and determining the first quality determined. And b.
Prohibiting further increase in the level of the probe signal when the parameter reaches a first
predetermined threshold. [Selected figure] Figure 2
Hearing aid with improved initialization of digital feedback suppression circuit parameters
[0001]
The present invention relates to a hearing aid, such as a hearing aid instrument, for example with
a digital feedback suppression circuit having parameters that are initialized during the fitting of
the hearing aid for a particular user.
[0002]
Feedback is a well-known problem in hearing aids and feedback suppression and canceling
systems are well known in the art (e.g., U.S. Pat. No. 5,619,580, U.S. Pat. No. 5,680,467 and U.S.
Pat. No. 6,498,858). Please refer to the specification).
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[0003]
Conventionally, digital feedback suppression circuitry is used in hearing aids to suppress
feedback signals from the receiver output.
During use, the digital feedback suppression circuit evaluates the feedback signal using, for
example, one or more digital adaptive filters modeled on the feedback path.
The feedback estimate from the digital feedback suppression circuit is subtracted from the
microphone output signal to suppress the feedback signal.
[0004]
The feedback signal may propagate from the receiver back to the microphone along an external
signal path outside the hearing aid housing and along an internal signal path inside the hearing
aid housing.
[0005]
External feedback (ie, propagation of sound from the hearing aid receiver to the microphone
along a path outside the hearing aid) is also known as acoustic feedback.
Acoustic feedback occurs, for example, when the ear form of the hearing aid is not perfectly
fitted to the wearer's ear, or, for example, when the ear form contains a canal or opening for
ventilation purposes. In both instances, sound can leak from the receiver to the microphone,
thereby causing feedback.
[0006]
Internal feedback can be caused by sound propagating through air in the hearing aid housing or
by mechanical vibrations of the hearing aid housing and components within the hearing aid
housing. For example, mechanical vibrations are generated by the receiver and transmitted to
other parts of the hearing aid, for example through the receiver mount. In some hearing aids, the
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receiver is flexibly attached to the housing. This reduces the transmission of vibrations from the
receiver of the hearing aid to the other components.
[0007]
WO 2005/081584 discloses a hearing aid with two separate digital feedback suppression
circuits for internal mechanical / acoustic feedback compensation and for external feedback
compensation.
[0008]
The external feedback path extends around the hearing aid.
Therefore, it is usually longer than the internal feedback path. That is, to travel from the receiver
to the microphone, the sound must propagate along the external feedback path a longer distance
than it would along the internal feedback path. Thus, when sound is emitted from the receiver,
the sound propagating along the external feedback path will arrive late at the microphone as
compared to the sound propagating along the internal feedback path. Thus, it is preferred that
the separate digital feedback suppression circuitry operate at the first and second time windows
respectively and that at least a portion of the first time window precedes the second time
window. Whether the first and second time windows overlap depends on the length of the
impulse response of the internal feedback path.
[0009]
While external feedback can vary greatly during use, internal feedback is more uniform and is
usually addressed during the manufacturing process.
[0010]
It is known that the correct initialization of the digital feedback suppression circuit is essential
for the effective suppression of feedback in the hearing aid.
In principle, adaptive filters automatically adapt to changes in the feedback path, but there is a
limit to the degree and accuracy of feedback path changes that the adaptive filter can track.
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However, accurate initialization of the digital feedback suppression circuit provides for a fast and
accurate modeling of feedback path response and effective feedback suppression during
subsequent operations by providing a starting point for adaptation close to the desired end
result. Connect. Initialization may occur anytime, during the fitting period, and perhaps every
time the user switches on the hearing aid.
[0011]
Typically, digital feedback suppression circuitry is initialized during the fitting of the hearing aid
for a particular user. The hearing aid is connected to the PC, the probe signal is transmitted to
the receiver, and based on the microphone output signal including the response to the probe
signal, the impulse response of the feedback path is evaluated. Typically, the probe signal is 10
seconds long, which is a high level that is difficult for the user to respond quickly. In order to
allow the user to adapt to the probe signal, the probe signal rises linearly on a logarithmic scale
from zero for one second prior to a 10 second constant level probe signal. The received
microphone output signals are sent to the PC and their respective impulse responses are
calculated. The PC then determines the parameters required by the digital feedback suppression
circuit, such as the filter coefficients of the fixed digital filter, and the initial filter coefficients of
the adaptive digital filter, so that the feedback path can be modeled.
[0012]
A hearing aid having more than one microphone, for example a directional microphone system,
may include a separate digital feedback suppression circuit for each microphone that is
initialized separately using the same probe signal.
[0013]
US Patent Publication No. 2002/0176584 discloses the initialization of a digital feedback
suppression circuit in which the level of the probe signal is adjusted according to the ambient
noise level.
The ambient noise level is determined based on the microphone output, and if the ambient noise
level is below a low threshold, the minimum probe signal is used. When the ambient noise level is
between the low threshold and the high threshold, the probe signal level is increased such that
the ratio of probe signal level to minimum probe level is equal to the ratio of ambient noise level
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to its threshold Be done. The probe signal level is not allowed to exceed the maximum value
selected for the user's comfort. If the ambient noise level exceeds the high threshold, the probe
signal level is limited to the maximum value.
[0014]
Traditionally, hearing aid users have complained about discomfort and distress during the
initialization process.
[0015]
In contrast, in recent years, open solutions have appeared.
According to the definition of hearing aids, hearing aids with a housing that does not block the
ear canal when placed in the intended working position in the ear canal are classified as "open
solutions". The term "open solution" is used for the following reasons. That is, a passage between
a portion of the ear canal wall and a portion of the housing where sound waves may escape from
behind the housing and between the tympanic membrane and the housing through this passage
to the user's surroundings It is used because it allows the passage. With an open solution, the
occlusion effect is reduced, preferably substantially eliminated.
[0016]
A standard sized hearing aid housing, typically with a high level of comfort and adapted to a
large number of users, is an example of an open solution.
[0017]
The open solution may be embodied in a feedback path with a long impulse response, since the
receiver output is not separated from the microphone input by the sealing of the ear canal.
This makes the feedback path relatively open and uses a long impulse response. This can further
increase the probe signal period required for feedback path evaluation.
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[0018]
Accordingly, it is desirable to provide a method for initializing digital feedback suppression
circuitry that reduces the user's discomfort during the initialization process.
[0019]
To address the above, a new initialization process is provided.
In this process, the level and length of the probe signal is maintained at the minimum value
necessary for proper initialization of the digital feedback suppression circuit. Initially, the probe
signal is increased linearly from an inaudible level, for example a low level such as zero, for
example by a logarithmic scale, during which the first quality parameter value is monitored.
When the first quality parameter value reaches a predetermined first threshold, the probe signal
is kept constant at the corresponding signal level, with which the second quality parameter value
is monitored. When the second quality parameter value reaches a predetermined second
threshold, the probe signal level is for example lowered again to an inaudible level, for example
turned off.
[0020]
The signal level is defined as, for example, the sound pressure level (SPL) generated by the
hearing aid at the sound pressure level (SPL) in front of the tympanic membrane or at the sound
input of the hearing aid microphone or a separate microphone that is not part of the hearing aid.
It is also good.
[0021]
The sound pressure level is a logarithmic scale of the rms sound pressure of the sound relative to
a reference value, and is measured in decibels (dB).
The commonly used reference sound pressure in air is 20 μPa (rms), which is usually considered
as the threshold for human hearing.
[0022]
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The sound pressure level is controlled by the signal level of the electronic input signal to the
receiver of the hearing aid, eg the rms value.
[0023]
There is no need to specify the resulting sound pressure level.
The maximum sound pressure level reached as a result is the correlation value of the first and
second thresholds of the first and second quality parameters, respectively.
[0024]
The sound pressure level may be determined by the selected frequency, the selected frequency
range, or the correlation value of the frequency. Alternatively, the sound pressure level may be
determined by substantially the entire frequency range of the probe signal.
[0025]
During quality parameter monitoring, the quality parameters in question are iteratively
calculated based on the microphone output signal, and the continuous values of these quality
parameters are compared to the associated first or second threshold.
[0026]
An increasing value of the first or second quality parameter may indicate an improvement in
quality in the microphone output signal.
This type of quality parameter starts at a low value and gradually increases. The respective first
or second threshold is reached when the quality parameter in question is above the respective
threshold.
[0027]
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In another type of quality parameter, decreasing values of the quality parameter indicate an
improvement in quality in the microphone output signal. This type of quality parameter starts at
high value and decreases gradually. Each threshold is reached when the quality parameter in
question falls below the threshold.
[0028]
For example, the first quality parameter may be related to the difference in the identified impulse
response of the feedback path. The ramping of the probe signal occurs when the identified
impulse response is sufficiently stable, ie, when the first quality parameter (which is a measure of
the difference in continuously identified impulse responses) is less than or equal to the first
threshold. It may stop.
[0029]
As another example, the first quality parameter may relate to the signal level at the hearing aid
microphone or an external microphone that is not part of the hearing aid. For example, the first
quality parameter may be equal to the rms value of the electronic output signal of the
microphone in question or its correlation value.
[0030]
Accordingly, a method of modeling a feedback path from a receiver to a microphone in a hearing
aid, comprising: recording a microphone output signal; and determining at least one parameter of
the feedback path based on the recorded microphone output signal, Sending an electronic probe
signal to the receiver to convert it to an acoustic probe signal output by the transmitting step, the
step of sending the probe signal to the receiver comprising: increasing the level of the probe
signal; Monitoring a first quality parameter value calculated based on the recorded microphone
output signal; and further increasing the level of the probe signal when the determined first
quality parameter reaches a predetermined first threshold. To ban A method is provided,
characterized in that the method comprises:
[0031]
The step of transmitting the probe signal comprises the steps of: monitoring a second quality
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parameter value calculated based on the recorded microphone output signal; and when the
determined second quality parameter reaches a predetermined second threshold, the receiver
Terminating the transmission of the probe signal thereto.
[0032]
The first quality parameter and the second quality parameter may be identical.
[0033]
The method may further include the step of evaluating the impulse response of the feedback
path.
[0034]
At least one of the first quality parameter and the second quality parameter may be a parameter
of an impulse response.
[0035]
The parameters of the impulse response are selected from the group consisting of: peak to peak
ratio of head and tail of impulse response; noise to noise ratio of head and tail of impulse
response; and peak to signal to noise ratio of impulse response You may
[0036]
In one embodiment, the digital feedback suppression circuit includes a fixed IIR filter and an
adaptive FIR filter.
The adaptive FIR filter coefficients may be updated based on the minimization of the minimum
mean square error.
Also, adaptive adaptive filters may be utilized during the initialization process.
After initialization, the filter continues its operation with fixed filter coefficients so that the filter
acts as a static filter.
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[0037]
The probe signal may be the longest sequence, eg, repeated 255 sample longest sequence, a
wideband noise signal, etc.
The use of the longest sequence avoids the occurrence of standing waves.
[0038]
The recorded microphone output signal, including the response to the probe signal, may be
uploaded to an external computer.
This external computer digital feedback suppression circuit to evaluate the feedback signal path
and transfer the determined parameters such as, for example, the filter coefficients of the fixed
digital filter and the adaptive digital filter to the digital feedback suppression circuit Is configured
to forward to.
[0039]
In one embodiment, the digital feedback suppression circuit includes an adaptive filter that is
adaptable during transmission of the probe signal to the receiver.
The initialization may end when the change of the filter coefficient becomes smaller than a
predetermined threshold that constitutes the second threshold. The change in filter coefficients
from one adaptation cycle to the next adaptation cycle constitutes the second quality parameter
value.
[0040]
According to the provided method, the user's discomfort uses a probe signal with a signal level or
amplitude that is large enough to make an assessment of the feedback path, but not larger than
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necessary. Reduced or eliminated.
[0041]
The determination of the required probe signal level starts the transmission of the probe signal
to the receiver from a low level, for example an inaudible level such as 0 dBSPL, and the impulse
response of the feedback path is sufficient for the determination of the required parameters This
may be done by gradually increasing the level of the probe signal until it is considered as good
quality.
This may be done, for example, by monitoring the change in the determined parameter of the
impulse response that constitutes the first quality parameter, and stopping the increase of the
level of the probe signal when the change is less than the first threshold. It will be.
[0042]
For example, the maximum allowable signal level and length of the probe signal may be applied
that is equivalent to the standard initialization signal level and length that would have been used
in a conventional initialization process.
[0043]
Similarly, the transmission of the probe signal at the determined constant level may be stopped if
the impulse response evaluation is considered to be of sufficient quality, and the length of the
probe signal may be as short as possible.
[0044]
The determined required level of the probe signal may vary depending on the type and model of
the hearing aid and the type of fitting (open / close).
[0045]
The rate of increase of the probe signal level may vary depending on the expected required
signal level and the predetermined time period set to reach the expected required signal level.
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The expected signal level may be, for example, 85 dBSPL for users without hearing impairment.
At the 85 dB SPL level, there is generally no discomfort experienced by a person with normal
hearing.
In general, users with hearing impairments use much higher initialization levels such as 102
dBSPL. In this case, the level may reach the maximum value of the output level of the device (e.g.
120 dBSPL), but is limited to a level that can limit the distortion due to receiver abuse.
[0046]
The calculation of the first and second quality parameters and the parameters of the digital
feedback suppression circuit may be performed by a computer external to the hearing aid, thus,
as is well known in the art, bi-directionally between the hearing aid and the external computer A
data communication link may be established. The external computer may receive the microphone
output signal and, according to the calculation of the first quality parameter and possibly the
calculation of the second quality parameter, eg start and stop signal generation by the probe
signal generator, probe signal generation Control of the probe signal generator may be
performed, such as the current signal level of the device output.
[0047]
The calculations and controls necessary to perform the initialization process may be shared
between the external computer and the hearing aid in various ways. For example, all necessary
tasks of the initialization process may be performed in the hearing aid if the signal processor has
sufficient computing power and memory to execute the corresponding program.
[0048]
Thus, a microphone for converting the input sound into an audio signal, a digital feedback
suppression circuit for modeling the feedback path of the hearing aid, a signal processor for
processing the compensated audio signal, and an audio processing of the processed signal A
receiver connected to the output of the signal processor for conversion to a signal; and a probe
signal generator for generating a probe signal to the receiver for conversion to an acoustic probe
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signal output by the receiver. The signal processor is further configured to record the
microphone output signal and to determine parameters of the digital feedback suppression
circuit based on the recorded microphone output signal, the signal processor increasing the level
of the probe signal. , With this Monitoring the value of the first quality parameter calculated
based on the recorded microphone output signal, and maintaining the level of the probe signal at
a constant level when the determined first quality parameter reaches a predetermined first
threshold value A hearing aid is provided, characterized in that it is further configured to:
[0049]
The signal processor as described above monitors the value of the second quality parameter
calculated based on the recorded microphone output signal, and the probe signal to the receiver
when the determined second quality parameter reaches a predetermined second threshold. May
be further configured to terminate the transmission of
[0050]
The signal processor may be further configured to evaluate an impulse response of the feedback
path.
[0051]
The digital feedback suppression circuit may constitute a feedforward control circuit.
[0052]
The digital feedback suppression circuit may constitute a feedback control circuit.
In this case, a microphone for converting the input sound into an audio signal, a digital feedback
suppression circuit for generating a feedback compensation signal by modeling the external
feedback path of the hearing aid, and subtracting the feedback compensation signal from the
audio signal. A subtractor for producing a feedback compensated audio signal, a signal processor
connected to receive the feedback compensated audio signal and configured to process the
compensated audio signal, and an audio signal processing the processed signal A receiver
connected to the output of the signal processor to convert the signal processor, and a probe
signal generator that generates a probe signal to the receiver to convert the acoustic probe signal
output by the receiver; Is Ma A hearing aid further configured to record a microphone output
signal and to determine parameters of the digital feedback suppression circuit based on the
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recorded microphone output signal, the signal processor increasing the level of the probe signal,
and And monitoring the value of the first quality parameter calculated based on the recorded
microphone output signal, and maintaining the level of the probe signal at a constant level when
the determined first quality parameter reaches a predetermined first threshold. A hearing aid is
provided, characterized in that it is further configured to:
[0053]
The above signal processor may comprise a digital feedback suppression circuit.
[0054]
The above and other features and advantages of the present invention will become more
apparent to those skilled in the art by describing in detail exemplary embodiments of the present
invention in conjunction with the accompanying drawings.
[0055]
FIG. 1 is a block diagram representing a conventional hearing aid system with one feedback
compensation filter.
FIG. 1 is a block diagram of a hearing aid system with both internal and external feedback
compensation filters.
FIG. 5 illustrates the distribution over time of prior art probe signal levels.
FIG. 4 is a diagram showing the distribution of probe signals according to the present method
and the prior art probe signal of FIG.
Fig. 2 is a schematic block diagram illustrating the operating principle of the method.
[0056]
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In the following, the invention will be described in more detail on the basis of the drawings
attached to the specification and showing an exemplary embodiment of the invention.
However, the invention may be embodied in different forms and should not be construed as
limited to the embodiments set forth herein. Rather, these embodiments are provided so that this
disclosure will fully and completely convey the scope of the present invention to those skilled in
the art.
[0057]
A block diagram of a conventional (prior art) hearing aid with feedback compensation filter 106
is shown in FIG. The hearing aid comprises a microphone 101 for receiving the input sound and
converting it into an audio signal. The receiver 102 converts the output from the hearing aid
processor 103 into, for example, an output sound that has been modified to compensate for the
user's hearing loss. Thus, the hearing aid processor 103 may include elements such as an
amplifier, a compressor and a noise reduction system.
[0058]
The feedback path 104 is shown as a dashed line between the receiver 102 and the microphone
101. Sound from receiver 102 may propagate along the feedback path to microphone 101,
which may lead to known feedback problems such as whistling noise.
[0059]
The (frequency dependent) gain response (or transfer function) H (ω) of the hearing aid (without
feedback compensation) is given by Where ω represents the (angular) frequency, F (ω) is the
gain function of feedback path 104, and A (ω) is the gain function provided by the hearing aid
processor 103.
[0060]
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[0061]
When feedback compensation filter 106 is enabled, filter 106 provides a compensation signal to
subtraction unit 105.
This causes the compensation signal to be subtracted from the audio signal provided by the
microphone 101 prior to processing by the hearing aid processor 103. By this, the transfer
function is as follows. Where F '(ω) is the gain function of the compensation filter 106. Thus, the
more accurately F '(ω) evaluates the true gain function F (ω) of the feedback path, the closer H
(ω) is to the desired gain function A (ω).
[0062]
[0063]
As mentioned above, feedback path 104 is typically a combination of internal and external
feedback paths.
[0064]
A hearing aid with separate digital feedback suppression circuitry is shown in FIG. 2 to
compensate for internal mechanical and acoustical feedback within the hearing aid housing and
to compensate for external feedback.
As mentioned above, the hearing aid comprises a microphone 201, a receiver 202 and a hearing
aid processor 203.
Internal feedback path 204 a is shown as a dashed line between receiver 202 and microphone
201. Furthermore, the external feedback path 204b between the receiver 202 and the
microphone 201 is shown (also in dashed lines). The internal feedback path 204a includes an
acoustic connection between the receiver and the microphone 201, a mechanical connection, or
a combination of both acoustic and mechanical connection. The external feedback path 204 b is
an acoustic connection between (mainly) the receiver 202 and the microphone 201. The first
compensation filter 206 is configured with the internal feedback path 204a as a model, and the
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second compensation filter 207 is configured with the external feedback path 204b as a model.
The first compensation filter 206 and the second compensation filter 207 provide separate
compensation signals to the subtraction unit 205, whereby both the feedback along the internal
and external feedback paths 204a, 204b are processed in the hearing aid processor 203
Canceled before it takes place.
[0065]
While the internal compensation filter 206 models the internal feedback path 204a, the internal
feedback path 204a is typically static or quasi-static. This is because the internal components of
the hearing aid do not substantially change their characteristics with respect to the transmission
of sound and / or vibrations over time. Thus, the internal compensation filter 206 may be a static
filter with filter coefficients derived from open loop gain measurements, which are preferably
made during the manufacture of the hearing aid. However, depending on the hearing aid, the
internal feedback path 204a may change over time, for example, if the receiver is not fixed and
can be moved around within the hearing aid housing. In this case, the internal compensation
filter preferably includes an adaptive filter that adapts to changes in the internal feedback path.
[0066]
The external compensation filter 207 is preferably an adaptive filter that adapts to changes in the
external feedback path 204b. These changes are usually much more frequent than the
aforementioned possible changes in the internal feedback path 204a, so the compensation filter
207 should adapt more quickly than the internal compensation filter 206.
[0067]
Because the length of the internal feedback path 204a is shorter than the length of the external
feedback path 204b, the impulse response of the external feedback path 204b is the impulse of
the internal feedback path 204a if the impulse responses of both feedback paths are measured
separately. Delayed compared to the response. The delay of the external feedback signal depends
on the size and shape of the hearing aid but usually does not exceed 0.25 ms (milliseconds).
Typical delays are, for example, 0.01 ms, 0.02 ms, 0.03 ms, 0.04 ms, 0.05 ms, 0.06 ms, 0.07 ms,
0.08 ms, 0.09 ms, 0.1 ms, 0.s. 11 ms, 0.12 ms, 0.13 ms, 0.14 ms, 0.15 ms, 0.16 ms, 0.17 ms,
0.18 ms, 0.19 ms, 0.2 ms, 0.21 ms, 0.22 ms, 0.23 ms, For example, 0.24 ms.
08-05-2019
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[0068]
The impulse responses of each of the inner and outer feedback paths 204a, 204b differ in signal
level. The reason is that the attenuation along the internal feedback path 204a usually reaches
the attenuation along the external feedback path 204b. Thus, the external feedback signal is
usually stronger than the internal feedback signal.
[0069]
In short, the internal and external feedback compensation filters 206, 207 differ in at least the
following three points. 1. Adaptive frequency required, Position of impulse response in time
domain, and Dynamic range of the impulse response.
[0070]
Thus, providing two compensating filters 206, 207 saves processing power as compared to
providing one single adaptive filter. This is because a single filter requires more filter coefficients.
Furthermore, differences in dynamic range can improve accuracy.
[0071]
Furthermore, by providing separate circuits for internal and external feedback compensation, the
new initialization process is improved for the same reason.
[0072]
The internal compensation filter 206 is preferably programmed during manufacture of the
hearing aid.
Thus, when the hearing aid is assembled, a model of the internal feedback path is evaluated. In
order to obtain a valid assessment of the internal feedback path 204, it is necessary to perform
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system identification of a hearing aid with a blocked external feedback path. One way to do this
is to place the hearing aid in a coupler (simulated ear) that provides the receiver with an
appropriate acoustic impedance, ie an impedance approximately equal to the impedance of the
wearer's ear. Any leaks, such as apertures in in-the-ear (ITE) hearing aids, must be sealed so that
all external feedback paths are eliminated. Hearing aids (and couplers) may also be placed in an
anechoic test box to remove acoustic reflections and noise from the surroundings. Next, a system
identification procedure such as open loop gain measurement is performed to measure F (w) (see
Equations 1 and 2 above). One way to do this is to have the device play back the MLS sequence
(longest sequence) at the output 202 and record it at the input 201. The internal feedback path
can be evaluated from the recorded feedback signal. The resulting filter coefficients for the model
are then stored in the device and used during operation of the hearing aid.
[0073]
FIG. 3 is a temporal distribution of prior art probe signal levels used for initialization of two
individual digital feedback suppression circuits in a hearing aid with a directional microphone
system including front and back microphones. During fitting, the hearing aid is connected to the
PC and the illustrated probe signal is transmitted to the receiver of the hearing aid. Based on the
microphone output signal including the response to the probe signal, the impulse response of the
feedback path of the front and rear microphones is evaluated. The illustrated probe signal rises
linearly on a logarithmic scale from zero level, for example, for one second, to allow the user to
adapt to the probe signal. Subsequently, the probe signal remains at a constant level for 10
seconds. Typically, the fixed level is of a size that confuses the user. The resulting front and rear
microphone output signals are sent to the PC and their respective impulse responses are
calculated. The PC then determines the necessary parameters of the respective digital feedback
suppression circuit, eg the initial filter coefficients of the adaptive digital filter, so that these
circuits can model the respective feedback path.
[0074]
FIG. 4 is a distribution of the prior art probe signal of FIG. 3 compared to the probe signal
generated according to the new initialization process. The new probe signal also initially rises
from a low level to a constant level. However, the constant level referred to here may be lower
than the constant level of the conventional probe signal. Also, the length of the probe signal at
the constant level may be shorter than that of the conventional probe signal at the constant level.
The new initialization process maintains the level and length of the probe signal at the minimum
required for the desired quality at initialization of the digital feedback suppression circuit.
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Initially, the probe signal rises from an inaudible level, for example a low level such as a zero
level, while the value of the first quality parameter is monitored. When the value of the first
quality parameter reaches a predetermined first threshold, the probe signal is kept constant at
the corresponding signal level, with which the value of the second quality parameter is
monitored. When the value of the second quality parameter reaches a predetermined second
threshold, the probe signal level is again reduced to an inaudible level, eg switched off.
[0075]
FIG. 5 schematically shows a hearing aid with digital feedback suppression circuit initialized
according to the new method. The probe signal is the longest sequence (MLS) signal, ie, the
longest generated in the MLS signal generator and output to an amplifier (ramp scale) with
controlled gain that is controlled over time as shown in FIG. It is a sequence (MLS) signal. The
feedback signal is received by the microphone and digitized, and blocks of signal samples are
stored in the frame accumulator. In the illustrated example, data blocks are transferred to the PC
for processing to extract impulse responses. The PC performs cross correlation between the
received signal and the probe signal and evaluates the impulse response. As an alternative to this,
the impulse response may be calculated by the signal processor of the hearing aid itself. The
quality of the impulse response is then evaluated by the PC in the example shown, but
alternatively by the signal processor of the hearing aid. The value of the first quality parameter is
calculated and compared to the first threshold. If the value of the first quality parameter does not
reach the first threshold, the probe signal level is increased. Otherwise, the signal level is
maintained at a constant level and a steady state measurement phase is entered. The value of the
second quality parameter is calculated and compared to the second threshold. If the value of the
second quality parameter does not reach the second threshold, a new data block is collected and
a new value of the second quality parameter is calculated. Otherwise, the initialization sequence
is ended. In the illustrated hearing aid, the PC calculates the corresponding parameter values of
the digital feedback suppression circuit and transfers the values to the hearing aid.
[0076]
The probe signal is subject to the maximum allowable signal level and length. This is equal to the
standard initialization signal level and length in the conventional initialization process.
[0077]
Quality parameters based on the impulse response of the feedback path are: Peak-to-peak ratio
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(PPR) of head and tail of impulse response Noise-to-noise ratio (NNR) of head and tail of impulse
response Peak-to-peak of impulse response It may be a signal noise ratio (PSNR) or the like.
[0078]
The impulse response may be extracted by the digital signal processor of the hearing aid.
The impulse response may be obtained by cross-correlating the MLS sequence with the reception
response. Although the DSP operates on a block basis, extracting the impulse response is a
computationally intensive process, and cross correlation can not be completed within one block.
Impulse response extraction should be done widely across many blocks.
[0079]
PPR is the ratio of the peak amplitude at the tail to the peak at the head of the impulse response,
expressed in dB. In the present application, the head and tail parts are defined as the first half
and the second half of the impulse response, respectively.
[0080]
NNR is the ratio of the noise level at the tail to the noise level at the head of the impulse
response, expressed in dB. In the present application, the head and tail parts are defined as the
first half and the second half of the impulse response, respectively. The level of noise is
calculated using the RMS value. In the configuration without a DC removal filter, variance can be
used to obtain similar results.
[0081]
PSNR is the ratio of root mean square (RMS) noise to signal peak and is expressed in dB. In the
present application, evaluation is made using the ratio of the RMS value of the last 64 samples of
the response to the peak amplitude of the extracted impulse response.
08-05-2019
21
[0082]
In the illustrated example, the new initialization process ends when both PPR and NNR exceed a
certain threshold. PSNR may also constitute a robust and reliable measure of quality.
08-05-2019
22
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