2. Related Art: A graphical representation of the"auditory area"of a young, adult subject with nor- mal, average hearing is shown in Fig. 1 in terms of the frequency and intensity ranges of sounds that are audible to the subject. As shown in the figure, the auditory area is bounded, in terms of audible frequencies, between about 20 Hz and 18 kHz.

In terms of intensity, as measured in decibels ("dB") relative to a reference sound pressure level ("SPL") of 20 micro-Pascals (20 dupa), the auditory area is bounded between two roughly upwardly-concave curves. The lower of these demarcates the"threshold of hearing"level, below which sounds are inaudible, and the upper curve demarcates the "threshold of feeling"level, at which sounds begin to produce a tickling sensation in the ear.

Sounds with intensities above the threshold of feeling can cause pain and permanent damage to the ear. For this reason, Underwriters Laboratories, Inc. , has promulgated a safety stan- dard for telephones (UL 1950, 3rd ed. ) that fixes an upper limit of 124 dB SPL on the inten- sity of sound that the earphone, or receiver, of a UL-certifiable telephone can produce, as shown in Fig. 1.

Subsumed within the auditory area of Fig. 1 are two overlapping regions of interest in the field of sound reproduction. The first of these is the area within which all sounds normally associated with the psycho-acoustic phenomenon of music occur, and the second, which is subsumed within the first, is the area within which all sounds normally associated with the psycho-acoustic phenomenon of speech occur.

It may be seen that the speech auditory area is considerably smaller than the music auditory area, and extends from about 100 Hz to about 6 kHz in frequency, and from about 25 dB to about 70 dB SPL in intensity. Moreover, it has been found that good speech intelli- gibility can be obtained with a minimal loss in sound quality, and with a substantial savings in transmission bandwidth, by limiting the telephonic bandwidth, or"pass band, "to about 3

kHz, with frequencies ranging from between about 200 Hz to 3.2 kHz. Thus, all telephonic speech sounds can be thought of as occurring within the rectangular"pass window"outlined Fig. 1 between 200 Hz and 3.2 kHz on the frequency axis, and between about 25 dB and 70 dB SPL on the intensity, or"volume"axis.

All persons suffer some deterioration in the acuity of their hearing with age, men more so than women, as is illustrated in Figs. 3 and 4, respectively, where the average dete- rioration in hearing in decibels at given frequencies with age in years are respectively plot- ted for men and women subjects. As may be generally inferred from Figs. 3 and 4, more than 80% of individuals who experience a loss in hearing suffer from a loss in the high fre- quency range, which, for telephonic hearing, is defined to extend from about 1.8 to 3.2 kHz.

Hearing losses may also be caused by other factors not related to age, including heredity, injury, and disease.

The telephonic speech pass window of Fig. 1 is reproduced in Fig. 2, which illus- trates the effect of a"flat, "60 dB hearing loss on telephonic communications. It may be seen that such a loss reduces a normal dynamic range of hearing DRl of about 124 dB to a residual dynamic range DR2 of about 64 dB, thereby effectively"masking"a substantial portion of the telephonic speech pass window (shown by the solid outline), which has a dy- namic range DRp of about 45-50 dB. Thus, nearly 80% of the telephonic speech received by an individual with such a hearing loss is inaudible to the individual, and therefore, lost. Ac- cordingly, one of the principal objects of any hearing-assistive device, such as a hearing aid or hearing assistive telephone, is to shift the received speech program window upwards such that its dynamic range DRp falls within the remaining dynamic range DRa of a hearing- impaired listener, as shown by the dashed outline in Fig. 2. In so doing, it may also be nec- essary to reshape the window, as with compression and limiting, such that it fits within that range.

The above object can be partially achieved in an amplified telephone set, such as the prior art amplified telephone set 10 illustrated schematically in Fig. 5 and discussed in more detail below, by simply adding linear gain, or amplification, to the received signal. How- ever, because of unpredictable levels of feedback in the system, including"line echo,""side tone, "and"acoustic echo, "there are at least two substantial limitations to this solution: First, since the telephone set forms part of a closed loop, not more than about 30 dB of gain can be added to the received signal without exceeding a total closed loop gain of unity, thereby inducing unstable operation, or"howl, "in the telephone, that is similar to that set up

in a public address ("PA") system when the microphone is placed too close to the output of the speaker. However, as may be seen from Fig. 2, as much as 45-50 dB of gain may be needed to completely shift the received signal's pass window into a hearing impaired user's remaining dynamic range of hearing, leaving a substantial deficit in the required amplifica- tion of 15-20 dB.

Second, the level of the hearing impaired user's transmitted signal that is fed back to the user in the form of line echo and side tone varies greatly and seldom matches that of the signal received by the user from a far-end talker. If the user adjusts the amplification of the set such that one of these signals is heard comfortably, the other signal may be either inau- dibly soft, i. e. , not heard, or heard uncomfortably loud and distorted. While these drawbacks can be addressed to a certain extent by so-called"echo suppression"techniques, i. e. , switching circuits that prevent the user from transmitting and receiving at the same time, this not only can add"switching artifacts"to the conversation, i. e. , annoying pops and clicks, but it also reduces the hearing-impaired user's communication from"full-duplex,"in which he can both transmit and receive simultaneously, to"half-duplex,"in which all side tone is ef- fectively eliminated. Since side tone provides important telephonic and voice modulation cues to the speaker, hearing-impaired persons need side tone as much or more than persons with normal hearing for effective communication.

Accordingly, a long felt and as yet unsatisfied need exists for a telephone set for hearing impaired users that affords the user with effective full duplex communication, and that enables as much 50 dB of gain to be applied to the received signal without inducing feedback howl in the set.

SUMMARY In accordance with the present invention, a telephone set is provided for hearing im- paired users that affords effective full duplex operation, and yet enables as much 50 dB of gain to be applied to the received signal without inducing feedback howl in the set.

One exemplary embodiment of the novel telephone set comprises a base part, and a handset or a headset portion incorporating a receiver and a transmitter. A portion of the sig- nal received by the receiver is coupled onto a signal transmitted by the transmitter as an acoustic echo signal. A hybrid located in the base portion couples the set to the two-wire analog tip and ring signals of the local loop, and also couples a portion of the transmitted signal back to the receiver as a conventional side tone signal. The received signal also in-

cludes a portion of the transmitted signal that is reflected back to the set from the local loop as a line echo signal.

Two codecs are arranged in the base portion to convert the respective received and transmitted signals of the telephone set between analog and digital formats. The base portion also includes a digital signal processor within which, among other functions, four filters are implemented. A first one of the filters is arranged to filter substantially all line echo and conventional side tone signals from the signal received by the receiver. A second one of the filters is arranged to filter substantially all acoustic echo signals from the signal transmitted by the transmitter. A third one of the filters is arranged to control the level of the filtered transmitted signal and then combine it, as a"replacement"side tone signal, with the filtered received signal to form a modified received signal. A fourth one of the filters comprises a receiver audio processor arranged to enhance the modified received signal by amplifying it, compressing its dynamic range, and limiting its maximum level"softly,"i. e. , without intro- ducing non-linearities into the signal.

In one exemplary preferred embodiment of the telephone, each of the first and sec- ond filters comprises an adaptive transversal filter. Each of these, in turn, comprises a sam- pler for sampling each of a reference signal and an object signal onto which at least a por- tion of the reference signal has been coupled as an echo signal. A computer generates a rep- lica of the echo signal from the reference signal, subtracts it from the object signal, and gen- erates an error signal corresponding to the difference between the echo signal and the replica signal. The computer includes an algorithm that operates on the error signal and previous values of the replica signal to successively adapt the replica signal in such a way that the error signal progressively approaches zero.

By canceling out the feedback elements of the closed loop, including the conven- tional side tone, and then feeding the filtered transmitted signal back to the receiver in a controlled manner as a replacement side tone, the filters enable more than 50 dB of gain to be added to the modified received signal without exceeding a total loop gain of unity, i. e. , without inducing unstable operation of the telephone set, and further, ensure that both the replacement side tone signal and the signal received from the far-end talker are heard by the user with an identical clarity and comfort.

A better understanding of the above and other details and advantages of the novel telephone set may obtained by a consideration of the detailed description below of an exem-

plary preferred embodiment thereof, particularly if such consideration is taken in conjunc- tion with the figures of the appended drawings.

DESCRIPTION OF THE DRAWINGS Figure 1 is a graphical representation of the auditory area of a subject having normal, average hearing, in terms of the frequency and intensity ranges of sounds that are audible to the subject and the areas subsumed therein within which music and speech occur; Fig. 2 is a graphical representation of a telephonic speech pass window, in terms of the frequency and intensity ranges of the sounds occurring therein, and showing the partial masking effect thereon of a flat, 60 dB hearing loss; Fig. 3 is a graphical representation of the average loss in hearing at various frequen- cies, in dB SPL, incurred by men with age, in years; Fig. 4 is a graphical representation of the average loss in hearing, in dB SPL, in- curred by women at various frequencies, with age, in years; Fig. 5 is a schematic, block diagram representation of a prior art amplified telephone set for hearing impaired users; Fig. 6 is a schematic, block diagram representation of a full duplex telephone set for hearing impaired users incorporating a digital signal processor in accordance with an exem- plary preferred embodiment of the present invention; and, Fig. 7 is a functional block diagram of the operation of the digital signal processor of the novel telephone set shown in Fig. 6.

DETAILED DESCRIPTION A prior art amplified telephone set 10 for hearing impaired users is illustrated sche- matically in Fig. 5, and is described in more detail in, e. g., U. S. Pat. Nos. 4,191, 861; 4,666, 120; and, Re. 33,309 to W. F. Walker, Jr. , et al. , which are commonly owned by the assignee hereof. The exemplary prior art telephone 10 illustrated has been simplified for discussion purposes by omitting certain conventional telephone set mechanisms used for signaling and switching, including, e. g., dialing, ringing, and switchhook mechanisms.

The conventional amplified telephone set 10 comprises a handset or headset 12, which includes an earphone, or receiver 14, for converting a received electrical signal into a corresponding sound, and a microphone, or transmitter 16, for converting a transmitted sound into a corresponding electrical signal. The set 10 also includes a hybrid 18, sometimes called a"two-to-four-wire converter, "for coupling the set, shown located to the right of the

dashed line 20, to the two"tip"and"ring"wires of the"local loop"22 of the telephone company service provider ("Telco"), located to the left of the dashed line.

The local loop comprises an analog DC circuit upon which an analog AC signal re- ceived from a remote, or far-end, talker (not shown), and an analog AC signal transmitted by the user to the far-end talker, are both superimposed, preferably sequentially, and not si- multaneously. The hybrid 18 thus comprises a multi-tapped transformer that inductively couples the signals transmitted and received by the set 10 to and from the local loop 22, re- spectively. The local loop 22 also includes a loading coil 24 to match the respective input and output impedances of the telephone set 10 and the local loop 22 to each other as closely as possible, and thereby minimize the amount of the respective transmitted and received signals that are reflected back, or"soaked-up, "by the wires in the loop. Nevertheless, de- pending on the location of the telephone set 10 within the local loop 22, a portion of the sig- nal transmitted by the transmitter 16 of the set is reflected back as a"line echo"superim- posed on the receive, or ring, signal.

The hybrid 18 in the telephone set 10 also couples a portion of the signal transmitted by the transmitter 16 of the set back onto the received signal as a"side tone"signal. The side tone signal comprises a form of intentional feedback to the user of a full duplex tele- phone set that enables the user to ascertain that a"connection, "or communication circuit, is open between the user and a far-end respondent, and also to modulate the volume and speech formatives of the user's voice for effective communication. Thus, the signal received by the telephone set 10 has two elements of feedback of the transmitted signal superimposed on it, viz., the line echo and side tone signals.

In the amplified telephone set 10 of Fig. 5, the received signal is passed first through a high pass filter 26 to provide a form of pre-emphasis to the high-frequency portions of the signal, then through a linear amplifier 28 to add an adjustable amount of gain to it. A low- pass filter, or"hard"limiter 30, is disposed at the output of the amplifier 28 to prevent the acoustic output signal of the receiver 14 from exceeding the 124 dB SPL volume level man- dated by the UL 1950 standard. The limiter 30 may be implemented by a pair of oppositely polarized diodes connected in parallel in the receiver circuit to act as a"clipper"at amplifier output signal levels exceeding a specified value. Such a limiter 30 is characterized as a "hard"limiter because, while it effectively prevents the output of the receiver from exceed- ing the maximum allowable value, the clipping of the amplified output signal introduces substantial harmonic distortions into the signal that can adversely affect its intelligibility.

Accordingly, it is desirable to operate the telephone set 10 at signal levels below that at which such clipping begins.

A third feedback path extant in the telephone set 10 consists of the acoustic path 32 between the receiver 14 and the transmitter 16 of the handset 12, represented by a coupler contained in a dashed outline in Fig. 5, by which a portion of the acoustic output signal pro- duced by the receiver is coupled onto the acoustic signal input of the transmitter as an "acoustic echo"signal. The acoustic echo signal does not ordinarily present a problem in conventional telephone sets, because the amount of the acoustic signal fed back to the transmitter 16 is small enough to preclude the total loop gain in the system from exceeding unity. However, in speakerphones and amplified telephones 10 such as that illustrated in Fig. 5, which incorporate substantial amplification of the received signal, this acoustic echo can present a problem in the form of unstable operation, or howl, in the telephone set when the gain added to the received signal is sufficient to cause the total loop gain in the set to exceed unity. This condition, in turn, provokes operation of the hard limiter 30, and effec- tive communication is thereby precluded until the loop gain is reduced below the critical level.

While the prior art telephone set 10 partially addresses some of the problems of hearing impaired users, it also includes certain limitations that need to be overcome to pro- vide such users with improved telephonic communication. One of these relates to the insta- bility of the telephone at high gain levels. As a practical limitation, no more than about 30 dB of gain can be added to the received signal without inducing howl in the set, although as much as 45-50 dB of amplification may be required to meet the needs of many hearing- impaired users.

Another limitation relates to the typical mismatch between the highly variable levels of the acoustic echo and side tone signals and that of the signal received from the far-end talker, the former typically exceeding the latter appreciably. If the amplification of the set is adjusted such that the side tone is heard comfortably, the received signal may be either in- audibly soft and not heard, or alternatively, heard uncomfortably loud and distorted, thereby precluding consistent, effective communication.

An exemplary embodiment of a telephone set 100 for hearing impaired users in ac- cordance with the present invention that overcomes the above and other problems of prior art amplified telephones is illustrated schematically in the functional block diagram of Fig.

6. As in the prior art telephone set 10 of Fig. 5, the exemplary telephone set 100 comprises a

handset or headset 112 incorporating a receiver 114 and a transmitter 116, as well as a hy- brid 118 for coupling the set to the tip and ring wires of the local loop (not illustrated). Also, as in the conventional set 10, the hybrid 118 couples a portion of the transmitted signal back onto the received signal as conventional side tone, and the receiver 114 and transmitter 116 define an acoustic path 132 by which a portion of the acoustic output signal produced by the receiver is fed back onto the acoustic signal input of the transmitter as an acoustic echo sig- nal.

The telephone set 100 may also incorporate other conventional telephone features, such as a dual-tone multi-frequency ("DTMF") dial pad 134 for dialing and data entry, a liquid-crystal display ("LCD") 136 for displaying a variety of information, such as tele- phone status, number dialed, caller ID, call waiting, programming steps, and the like, a manual volume control device 138, such as a rotatable or slidable knob, for the user's con- venience in adjusting the volume of the receiver 114 acoustic output signal, and a micro- controller 140 for providing integrated control over the various functions and features of the set. The telephone set 100 may also include a quantity of erasable programmable memory ("EPROM") (not illustrated) for storage of frequently called numbers, as well as telephone set configuration data.

Unlike the conventional amplified telephone set 10 of Fig. 5, which is completely "analog"in nature, the exemplary telephone set 100 also comprises first and second codecs 142,144 each comprising an analog-to-digital converter ("A/D") 146 and a digital-to-analog converter ("D/A") 148, for converting the respective received and transmitted signals of the set between analog and digital formats, and a digital signal processor ("DSP") 150 for proc- essing those signals in the manner described below to achieve many of the advantages of the set.

The codecs 142,144 may comprise conventional voice codecs, such as Texas In- struments, Inc. TLV 320AIC-series codecs, and may be incorporated on a single chip in a so-called"dual codec"of a type found in some digital stereo receivers. The DSP 150 may comprise, for example, a Texas Instruments, Inc. TMS320-series DSP. The DSP 150 may be embodied in a custom-designed, application specific integrated circuit ("ASIC"), a field- programmable gate array ("FPGA"), or a"hybrid"between the two types that is partly cus- tom-designed and partly field-programmable.

As illustrated in Fig. 6, the DSP 150 of the telephone set 100 is programmed to im- plement, inter alia, four signal processing functions on the received and transmitted signals,

viz., a first filter 152 arranged to filter substantially all line echo and conventional side tone signals from the signal received by the receiver 114, a second filter 154 arranged to filter substantially all acoustic echo signals from the signal transmitted by the transmitter 116, a third filter 156 arranged to"condition,"i. e. , to control the level of, the filtered transmitted signal such it matches that of the filtered received signal and then combine the controlled signal with the filtered received signal to form a modified received signal, and a fourth, re- ceiver audio processor filter 158 arranged to enhance the modified received signal by, inter alia, amplifying, compressing, and"softly"limiting (i. e. , without distorting) the modified received signal, prior to its introduction into the receiver 114.

A more detailed description of the operation of the first, second and third filters 152, 154 and 156 is given below in connection with Fig. 7, and a more detailed description of the operation and advantages of the receiver audio processor filter 158 is given in the above- referenced, commonly owned, co-pending application Ser. No. , filed , the teachings of which are incorporated herein by this reference.

With reference to Fig. 6, the processing applied to the signals received and transmit- ted by the telephone set 10 is summarized as follows: The analog ring signal, which com- prises the analog signal received from a far-end talker, plus the analog line echo signal, is coupled into the set by the hybrid 118, which also couples the conventional analog side tone signal onto the ring signal, as described above. These combined analog received signals are then sampled, preferably at a rate of 8 kHz or better to avoid aliasing, by the A/D converter 146 in the first codec 142, and quantized into a corresponding digital signal. The digitized received signal is then processed, or filtered, by the first filter 152 to remove substantially all line echo and conventional side tone signals from the signal.

In a similar fashion, the analog signal transmitted by the transmitter 116, comprising the acoustic signal generated by the user's voice, plus the acoustic echo signal coupled from the receiver 114, is converted to a corresponding digital signal by the A/D converter 146 in the second codec 144. This digitized transmitted signal is then filtered by the second filter 154 to remove substantially all of the echo signal contained therein. The filtered transmitted signal is then"conditioned"by the third filter 156 in the manner described below such that its level closely matches that of the filtered received signal, and is then combined with the latter signal, as a"replacement side tone"signal, to form a modified received signal. The filtered transmitted signal is also converted to an analog format by the D/A converter 148 in

the first codec 142 and coupled onto the tip wire of the local loop by the hybrid 118 as the analog signal transmitted by the set 100.

The modified received signal is filtered by the receiver audio processor filter 158, which amplifies, compresses, and limits the signal to enhance its audibility and clarity to the hearing impaired user. This enhanced received signal is then converted to an analog signal by the D/A converter 148 in the second codec 144, passed through a linear amplifier 160, which in the particular embodiment described herein, serves primarily to match the output impedance of the D/A converter to the input impedance of the receiver 114, and is then con- verted by the receiver to an enhanced acoustic signal heard by the hearing impaired user's ear.

It may be seen from the foregoing that, by filtering substantially all of the acoustic echo feedback from the transmitted signal, the telephone set 100 is effectively converted from a"closed-loop"system to an"open-loop"system, such that more than 50 dB of"open loop"gain can be added to the modified received signal by the receiver audio processor fil- ter 158 without inducing feedback instability in the system. Additionally, effective full du- plex operation of the set 100, in which both the replacement side tone signal and the signal received from the far-end talker are heard by the user with the same clarity and comfort, is achieved by first filtering substantially all of the line echo and conventional side tone signals from the received signal, and then replacing the conventional side tone signal with a condi- tioned side tone signal as described above.

The operation of the first, second, and third filters 152,154 and 156 of the DSP 150 is described in more detail in connection with the functional block diagram of Fig. 7. In the particular exemplary preferred embodiment illustrated, each of the first and second filters 152,154 comprises an"adaptive transversal filter"of a type described by D. Messerschmitt et al. in Texas Instruments Application Report SPRA129, 1989, the teachings of which are incorporated herein by this reference.

In Fig. 7, each of the relevant signals is shown in its discrete-time, or sampled, form f (i), i. e., after conversion by the respective A/D converters 146 of the codecs 142 and 144 (not shown in Fig. 7), where i is an integer multiplier of the period T of the samples, e. g., 0.125 milliseconds (ms) for the exemplary 8 kHz sampling rate. It may be noted from the above discussion that the line echo and conventional side tone signals that are respectively coupled onto the signal received from a far-end talker, p (i), are both functions of, and hence, highly correlated with, the transmitted signal T (i), and accordingly, have been combined in

Fig. 7 into a single line echo and side tone signal eb (i) The acoustic echo signal coupled onto the transmitted signal is likewise highly correlated with the enhanced received signal R (i), is represented by ea (i).

Assuming a linear, time-invariant (or slowly time-varying) system, the respective echo signals ea (i) and eb (i) that are coupled onto the respective"object signals"p (i) and t (i) can be represented as the convolution of a respective"reference signal"to which they are correlated, viz., T (i) and R (i), and the respective discrete-time representation hka and hkb of the unit impulse response of the respective echo paths 162 and 164 over which the echo sig- nals are coupled to their respective object signals, as follows: N consecutive values of the functions R (i-k) and T (i-k) can be generated by applying the respective signals R (i) and T (i) to a delay line having N taps, as shown in Fig. 7, and if the corresponding N values of the respective impulse responses hka and hkb of the respective echo paths 162 and 164 are known, then exact replicas ea (i) and eb (i) of the respective echo signals ea (i) and eb (i) can be computed, which, when subtracted from their respective object signals, leave remaining signals from which the respective echoes have been cancelled ex- actly. Thus, as shown in Fig. 7, the remaining, or filtered, transmitted signal is given by T (i) = t (i) + [ea (i)-ea (i) ], and the remaining, or filtered, received signal is given by P (i) = p (i) + [eb (i)-eb (i) ]. Further, it will be seen that the functions [ea (i)-ea (i) ] and [eb (i)-eb (i) ] re- spectively define error signals Fa (i) and Fb (i) that approach zero as the respective replica sig- nals ea (i) and eb (i) more exactly replicate their respective echo signals ea (i) and eb (i).

The number of taps in the delay line is determined by the duration of the impulse response of the respective echo path, i. e., the time span over which it has a non-zero value.

This is typically on the order of 2 to 4 ms, corresponding to from about 16 to 32 tap posi- tions. However, echo paths incorporating significant delays require more taps for effective filtering. Thus, 128 taps are required to accommodate signals with delays of up to 16 ms, and in general, the more taps required, the more"computation-intensive"the filtering op- eration becomes.

The values of the respective impulse responses hka and hkb of the respective echo paths 162 and 164 are generally not known in advance, and further, vary slowly with time.

However, they can be approximated with respective"filter coefficients"ak and bk, which, when progressively"adapted"to more accurately represent the actual values of the respec- tive impulse responses hka and hkb, produce progressively more accurate replicas ea (i) and eb (i) of the echo signals ea (i) and eb (i).

An algorithm for progressively adapting the filter coefficients can be derived by minimizing the mean-squared error between the respective echo signal and its replica at a time when the respective signal components t (i) and p (i) of the respective object signals are respectively zero, i. e. , when the user and the far-end respondent are respectively silent. One known recursive algorithm for adapting the respective coefficients in terms of a preceding value of the coefficient and the respective error signal e (i), called the"least-mean-square" (LMS) algorithm, or the"stochastic-gradient algorithm, "is given by ak (i +1) = ak (i) + 2ß [£a (i) + R (i-k)], and bk (i +1) = bk (i) + 2 (3 [sb (i) + T (i-k)], respectively, where Sa (i) and Fb (i) are the respective error signals discussed above, and (3 is a constant proportional to the size of the"step"used in the approximation, where larger step sizes produce faster convergence but potentially larger residual echoes, and smaller step sizes result in slower convergence but smaller residual echoes. As will be appreciated by those of skill in the art, there are many other algorithms than can be advantageously applied to adapt the coefficients ak and bk of the filters 152 and 154, depending on the particular cir- cumstances at hand.

As will also be appreciated, it is possible that the respective transmit and receive sig- nal components t (i) and p (i) of the respective object signals of the echo-canceling first and second filters 152 and 154 will both be non-zero simultaneously, which occurs when both the user and the far-end respondent inadvertently talk at the same time, i. e. , short, so-called "doubletalk"periods. When this occurs, it is necessary to"freeze"the adaptation of the re- spective filter coefficients ak and bk and temporarily use previously stored values of the co- efficients to avoid divergence of the adaptive algorithm. This is effected by incorporating a "speech-detector"algorithm in the filters 152,154 that detects simultaneous non-zero t (i) and p (i) signals, and instructs the respective filters to suspend the computation of filter coef- ficients and to use the previously stored values thereof during the periods that this condition exists.

As shown in Fig. 7, the third filter 156 conditions the filtered transmitted signal such that its level matches that of the filtered received signal in the following manner. During

normal use, the hearing impaired user sets the volume level of the telephone set 100, i. e. , the amount of gain added to the modified received signal m (i) by the receiver audio processor filter 158, such that the enhanced received signal R (i) is heard at a desired level of comfort.

During this adjustment, the user is silent, so that the filtered transmitted signal T (i), and hence, the replacement side tone signal aT (i), are both zero. The modified received signal m (i) therefore consists entirely of the signal received from the far end talker, p (i). Con- versely, when the user is speaking, the far-end talker is silent, so that the signal received from the far end talker p (i), and hence, the filtered received signal P (i), are both zero, and the modified received signal m (i) then consists entirely of the replacement side tone signal aT (i).

Thus, after the user has set the volume of the enhanced received signal R (i) to the desired level, the third, or replacement side tone conditioning filter 156 functions to monitor the level of the output signal R (i) from the receiver audio processor 158, and stores the level of that signal when t (i) is zero and p (i) is non-zero. Thereafter, when p (i) is zero and t (i) is non-zero, the side tone conditioning filter 156 compares the level of the signal R (i) with that of the previously stored values and either adds or subtracts the amount of gain or attenuation a to the filtered transmitted signal T (i) necessary to maintain the level of the enhanced re- ceived signal R (i) at the comfort level initially set by the user. The same speech detector al- gorithm used above to detect doubletalk can also be used to detect the status of p (i) and t (i), and those of skill in the art will recognize this conditioning of the replacement side tone sig- nal aT (i) as a form of"automatic gain control" ("AGC") that is applied only to the filtered transmitted signal T (i).

Those of skill in the art by now will also appreciate that many modifications and variations are possible in the apparatus and methods of the present invention without de- parting from its scope and spirit. For example, it is possible to filter noise from the received signal using known digital processing techniques in the DSP, thereby improving the signal to noise ratio of the receiver. Accordingly, the scope of the present invention should not be limited to that of the particular embodiments described and illustrated herein, as these are merely exemplary in nature, but rather, should be commensurate with that of the claims ap- pended hereafter and their functional equivalents.