Electrical audio signal processing systems and devices – Hearing aids – electrical
Reexamination Certificate
2000-01-05
2004-05-04
Kuntz, Curtis (Department: 2743)
Electrical audio signal processing systems and devices
Hearing aids, electrical
C381S316000, C381S320000, C381S321000
Reexamination Certificate
active
06731767
ABSTRACT:
FIELD OF THE INVENTION
The present invention relates to the field of devices and methods for processing sound and in particular to a processor for improving the speech perception and comfort of a hearing impaired user. However, while the invention is suited for use with hearing impaired people it will also find application in other communication areas.
BACKGROUND
In general the effects of hearing impairment are characterised by the undesirable conditioning of a sound signal, for example spoken words, along a listener's hearing chain so as to result in attenuation and often distortion of the signal.
Relatively simple linear gain hearing aids, for example fixed gain aids, have been successful in amplifying sounds to make them audible and recognisable. One problem with fixed gain aids however is that they are usually not suitable for use over a wide range of sound frequencies and levels. For example, when using a fixed gain aid the listener often finds that some sounds are inaudible, that is below hearing threshold, while others are at, or above, the loudness discomfort level, (LDL). Such a problem is especially prevalent when the listener is a person with a narrow dynamic range between the threshold and LDL levels.
Multi-band compression schemes attempt to overcome the problems of narrow dynamic range by adapting the gain of the aid in response to changes in the input sound level within a number of frequency bands, that is, they make use of a non-linear compression scheme. However, non-linear compression schemes introduce distortions into the output signals which reduce speech intelligibility. Hearing aids incorporating multi-band compression schemes are also difficult to fit and may require a lengthy investigation of the subject's hearing response.
One type of multi-channel hearing aid is the subject of U.S. Pat. No. 5,687,241 to Ludvigsen. In that document there is described a multi-channel hearing aid which splits an input signal into a number of parallel, filtered channels. The filtered input signals are each monitored by a percentile estimator and on the basis of control signals generated by the percentile estimators the gain of each of the filtered signals is adjusted. The filtered, gain adjusted signals are then recombined, amplified and converted to an acoustic signal.
A problem with the aid of U.S. Pat. No. 5,687,241 is that the percentile estimators must be capable of accommodating large swings in the amplitude of the signal being monitored. Consequently in a digital implementation considerable processing power is required in order to undertake the percentile estimation calculations.
A further problem that arises during the operation of multi-channel hearing aids is that fast transient signals having magnitudes exceeding the maximum comfort level may arise. Typically such transients occur in only a small number of channels at a particular time however in order to prevent discomfort to the user of the aid the general prior art approach has been to reduce the total power output of the aid. While such an approach prevents discomfort it causes undesirable distortion of the signal in channels unaffected by fast transient signals.
Single channel automatic gain control (AGC) hearing aids operate to reduce the gain at all frequencies in the event that the level of a sound should reach a predetermined point. While such hearing aids prevent the sound from reaching the subject's LDL they also attenuate some frequency components of the speech signal to such an extent that the intelligibility of the speech is reduced.
In summary, prior art hearing aids have associated with their use a variety of problems. Such problems range from inappropriate compression of signal, which causes undue signal distortion, to onerous processing requirements which make the aids expensive and difficult to implement.
In light of the prior art it is an object of the present invention to provide an apparatus which, in the presence of an ambient sound signal, generates a transformed sound signal which conforms to predetermined amplitude requirements within a range of audible frequencies.
It is a further object of the invention to provide a means whereby fast transient signals may be suppressed, in order to prevent discomfort to the user of a multi-channel hearing aid, without introducing signal distortion into channels unaffected by said transient signals.
SUMMARY OF THE INVENTION
According to a first aspect of the present invention there is provided a method for processing an ambient sound signal including the steps of
a) generating an input spectrum comprising a plurality of frequency components corresponding to said signal;
b) multiplying each of said frequency components by a corresponding one of a plurality of gain values to produce a plurality of adjusted frequency components;
c) determining distribution values characteristic of the amplitude distribution of each of the plurality of adjusted frequency components over a period of time;
d) setting said gain values on the basis of comparisons between said distribution values and a plurality of hearing response parameters.
According to a further aspect of the present invention there is provided an apparatus for processing an ambient sound signal including:
a) a frequency analysis means arranged to generate an input spectrum comprising a plurality of frequency components corresponding to said ambient signal;
b) a magnitude adjustment means coupled to the frequency analysis means and arranged to adjust the magnitude of each of said frequency components to produce an output spectrum comprising a plurality of adjusted frequency components corresponding to an adjusted sound signal related to the ambient sound signal;
c) a distribution estimation means responsive to said plurality of adjusted frequency components and arranged to generate distribution values characteristic of the amplitude distribution of each of the said plurality of adjusted frequency components over period of time; and
d) a comparison means coupled to the distribution estimation means and arranged to perform comparisons of said distribution values with hearing response parameters, said comparison means controlling said magnitude adjustment means on the basis of said comparisons.
Preferably, the frequency analysis means, the magnitude adjustment means, the distribution estimation means and the comparison means are implemented by a programmed microprocessor coupled to memory storage means.
Preferably, the apparatus further includes a signal conversion means by which the output spectrum is converted to a sound signal for presentation to a human listener.
Alternatively the output spectrum might be further processed by a further signal processor, such as for example, a cochlear prosthesis.
Preferably, the parameters characteristic of the hearing response include maximum comfortable level, threshold level, and optimum audibility level for each of the plurality of frequency components.
It will be realised by the skilled addressee that, because the magnitude adjustment means of the present invention is controlled by a comparison means which performs a comparison on the output spectrum, rather than the input spectrum, the above apparatus operates to ensure that the adjusted sound signal conforms to predetermined amplitude requirements across a range of audible frequencies, thereby achieving at least the object of the invention.
Preferably, the frequency analysis means, the magnitude adjustment means, the distribution estimation means, and the comparison means, referred to above are implemented by a programmed microprocessor. Nevertheless it will be realised that other implementations are possible, for example the invention could be implemented using dedicated hardware rather than a microprocessor, or even in a substantially analog form, the construction of such implementations will be apparent to those skilled in the art in light of the following description of a preferred embodiment.
According to a final aspect of the invention there is provided a multi-channel hearing aid having a pl
Blamey Peter John
James Christopher John
Martin Lois
McDermott Hugh Joseph
Wildi Konrad
Gottlieb Rackman & Reisman P.C.
Nguyêñ Tuân D
The University of Melbourne
LandOfFree
Adaptive dynamic range of optimization sound processor does not yet have a rating. At this time, there are no reviews or comments for this patent.
If you have personal experience with Adaptive dynamic range of optimization sound processor, we encourage you to share that experience with our LandOfFree.com community. Your opinion is very important and Adaptive dynamic range of optimization sound processor will most certainly appreciate the feedback.
Profile ID: LFUS-PAI-O-3235474