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In article <1990Oct16.163935.1954@ux1.cso.uiuc.edu> sinistar@mrcnext.cso.uiuc.edu (Jeff O'Hare) writes:
>Hi.  I was wondering if anyone has heard of any dsp hearing aids.  I was think-
>ing about it last night, and it seems like it could be a great idea.  Real-time
>digital equalization could maybe give users more natural hearing, correcting
>the natural degradation of the ear.  I have heard (no pun intended) that hear-
>ing loss of high frequences is complete loss.  Is this true, and if so, what
>is the approximate rolloff rate?  With the DSP, such possibilities as harmon-
>ization could be used to lower sounds an octave so higher frequencies could
>still be heard, just an octave lower.  Well, has anyone else thought about
>this, or seen any made yet?  How about some dsp chips small enough to fit in
>a hearing aid?  The 96002 is pretty large I think...
>	Jeff

There has been an enormous amount of interest in this area and a large amount
of research.  There is/was at least one device already on the market (but it
has not been well-received by clinicians as far as I've heard). There is
research on DSP hearing aids in progress at CUNY, at the Central Institute
for the Deaf, at the Research Laboratory of Electronics at MIT, and many
other places. Lots of different kinds of signal processing have been tried
including amplitude compression (hearing impaired people often have a very
restricted dynamic range below which they cannot hear anything and above which
things are painfully loud), spectral peak enhancement (amplifying only the
most important parts of the instantaneous speech spectrum via adaptive
filtering), frequency compression as Jeff suggested above to fit more of the
speech signal into the reduced auditory bandwidth, frequency dependent
amplification to compensate for the attenuation characteristics of the
hearing loss, and other schemes. To date, about the only one of these that
always seems to provide clear improvements in speech reception
for hearing impaired people is frequency dependent amplification. Some
amplitude compression is probably desirable (some of the time). The more
complex schemes have tended to give sort of spotty results--sometimes
promising, other times not.

What all this means is that so far the one clear advantage of DSP hearing aids
is their adaptability: the same piece of hardware can be programmed to the
needs of many different users quite easily (given the necessary programming
hardware). By contrast, analog hearing aids that shape the speech spectrum to
compensate for the loss of the wearer must be prescribed and built (or at
least tweeked) on a one-at-a-time basis.  While this is potentially an
important advantage, the economics of it have not worked out favorably yet.
Analog aids are and will probably remain much cheaper for some time.

Many people working in this area (including me) think that effective
speech enhancement schemes will probably be based on signal processing
that imitates what people do with their speech when they try to speak
clearly (as opposed to their ordinary casual speech). We just haven't
been clever enough to figure out how to do that automatically yet. Any
new ideas out there?

--
Tim Bunnell
<bunnell@udel.edu>

In article <33682@nigel.ee.udel.edu>, bunnell@udel.edu (H Timothy Bunnell) writes:
> In article <1990Oct16.163935.1954@ux1.cso.uiuc.edu> sinistar@mrcnext.cso.uiuc.edu (Jeff O'Hare) writes:
> >Hi.  I was wondering if anyone has heard of any dsp hearing aids.  I was think-

Hmmm. Why build a switched capacitor filter, or fancy processing for 
gain stage controls, etc., when straight forward analog ciruits will
do the job efficiently and compactly? It's only audio range signals anyway,
but it needs to be handled in real-time. Why ask for design sophistication
when you don't need it? 

I also had a fascinating conversation with an audiologist here about
different compression schemes. I won't go into details since I'm an EE, not
an audiologist, but one scheme that works surprisingly well is clipping:
map a 60 dB range speech signal into a 30 dB comfortable range for the
subject by clipping the upper 30 dB right off. Some guy called Lickliter
documented this technique in the 40's, and found that as long as the 
frequencies, relative intensity of them, and timing information was
preserved, speech was still comprehensible.

What it does to music, you don't even want to think about!

I believe that more effective processing for speech enhancement will
be based on what humans actually do, rather than what our mathematical
analyses insist is most efficient or powerful. I'm not certain, however,
that we understand enough of how people *hear* and process sound to 
approach speech enhancement from the speech production side.

Note: this is my personal opinion as an educated EE, not as an audiological
expert of some sort.

-- 
			Nico Garcia
			Designs by Geniuses for use by Idiots
			eplunix!cirl!raoul@eddie.mit.edu

There was a neat project going on when I was in college last year (I 
didn't get the chance to work on it). The basic idea was to apply pattern 
recognition techniques to the input audio signal, and recognize the voiced 
plosive phonemes (which are very troublesome in certain types of hearing 
imparments involving large high frequency attenuations). The recognized 
phonemes would then be substituted with "in-band" phonemes, synthesized or 
taken from a foreign language. Gradually the user would map the new, 
in-band phonemes to the lost voiced plosives.  Don't know how far it's 
come or if anyone other than the U. of Waterloo is working on it; know of 
any similar efforts?

In article <10810@goofy.Apple.COM> zarko@apple.com (Zarko Draganic) writes:
>There was a neat project going on when I was in college last year (I 
>didn't get the chance to work on it). The basic idea was to apply pattern 
>recognition techniques to the input audio signal, and recognize the voiced 
>plosive phonemes (which are very troublesome in certain types of hearing 
>imparments involving large high frequency attenuations). The recognized 
>phonemes would then be substituted with "in-band" phonemes, synthesized or 
>taken from a foreign language. Gradually the user would map the new, 
>in-band phonemes to the lost voiced plosives.  Don't know how far it's 
>come or if anyone other than the U. of Waterloo is working on it; know of 
>any similar efforts?

I have heard of similar approach.  In the 70's (or 60's), one hearing aid
company designed an aid which 'mapped' frequencies that a user could not
hear into ones they could hear.  I cannot remember the name of the aid.  
I was told that the hearing aid was not comerically successful because of
problems associated with training the users to understand the newer 
frequencies.  Also, distingushing the frequencies apart was difficult.
Does anybody know more about this?  I would like to find out since the reasons
for the aids lack of commercial success would be interesting.
Thank you,
-------------------------------------------------------------
Alan Goldschen                       e-mail: ajg@seas.gwu.edu
Department of EE and CS
George Washington University
801 22nd Street N.W.
Washington, D.C. 20052

In article <952@eplunix.UUCP> raoul@eplunix.UUCP (Nico Garcia) writes:
>I also had a fascinating conversation with an audiologist here about
>different compression schemes. I won't go into details since I'm an EE, not
>an audiologist, but one scheme that works surprisingly well is clipping:
>map a 60 dB range speech signal into a 30 dB comfortable range for the
>subject by clipping the upper 30 dB right off.

Actually, one-bit speech is intelligible: take your digital speech,
map all positive numbers into the same value, and all negative numbers
into the same value (with opposite sign).  Sounds like hell but you can
understand it.

>I believe that more effective processing for speech enhancement will
>be based on what humans actually do, rather than what our mathematical
>analyses insist is most efficient or powerful. I'm not certain, however,
>that we understand enough of how people *hear* and process sound to 
>approach speech enhancement from the speech production side.

People who do real speech processing fight this battle from both ends,
and while there are many mysteries it's not as black an art as you
suggest that it is.  For example, in speech compression your ears really
don't care if your method has the minimum mean-square error; however, if
you apply a certain type of frequency weighting before you attempt to
minimize the error it works quite well.  To produce low-bit-rate speech
of good quality you have to know a lot of DSP and ALSO know a good deal
about the features of the human auditory system.


--
--
Joe Buck
jbuck@galileo.berkeley.edu	 {uunet,ucbvax}!galileo.berkeley.edu!jbuck