Abstract:
There is provided a method for adjusting a system for providing hearing assistance to a user ( 101 ), the system comprising a microphone arrangement ( 26 ) for capturing audio signals, a transmission unit ( 102 ) for transmitting the audio signals via a wireless link ( 107 ) to a receiver unit ( 103 ) worn by the user, a gain control unit ( 126 ) located in the receiver unit for setting the gain applied to the audio signals, and means ( 38 ) worn at or in a user&#39;s ear ( 39 ) for stimulating the hearing of the user according to the audio signals from the receiver unit ( 103 ), said method comprising: generating test audio signals, transmitting said test audio signals at a pre-defined level from the transmission unit via the wireless link to the receiver unit and stimulating the user&#39;s hearing with said test audio signals via said stimulating means; simultaneously transmitting gain control commands from the transmission unit to the gain control unit in order to selectively change the gain set by the gain control unit; repeating these steps until an optimum value of the gain set by the gain control unit has been determined; and transmitting a store command from the transmission unit to the receiver unit in order to store that determined optimum value of the gain.

Description:
BACKGROUND OF THE INVENTION 
   1. Field of the Invention 
   The present invention relates to a method for adjusting a system for providing hearing assistance to a user; it also relates to a corresponding system. In particular, the invention relates to a system comprising a microphone arrangement for capturing audio signals, a transmission unit for transmitting the audio signals via a wireless audio link from the transmission unit to a receiver unit, and means worn at or in the user&#39;s ear for stimulating the hearing of the user according to the audio signals received by the receiver unit. 
   2. Description of Related Art 
   Usually in such systems the wireless audio link is an FM radio link. According to a typical application of such wireless audio systems the receiver unit is connected to or integrated into a hearing instrument, such as a hearing aid, with the transmitted audio signals being mixed with audio signals captured by the microphone of the hearing instrument prior to being reproduced by the output transducer of the hearing instrument. The benefit of such systems is that the microphone of the hearing instrument can be supplemented or replaced by a remote microphone which produces audio signals which are transmitted wirelessly to the FM receiver and thus to the hearing instrument. In particular, FM systems have been standard equipment for children with hearing loss in educational settings for many years. Their merit lies in the fact that a microphone placed a few inches from the mouth of a person speaking receives speech at a much higher level than one placed several feet away. This increase in speech level corresponds to an increase in signal-to-noise ratio (SNR) due to the direct wireless connection to the listener&#39;s amplification system. The resulting improvements of signal level and SNR in the listener&#39;s ear are recognized as the primary benefits of FM radio systems, as hearing-impaired individuals are at a significant disadvantage when processing signals with a poor acoustical SNR. 
   Most FM systems in use today provide two or three different operating modes. The choices are to get the sound from: (1) the hearing instrument microphone alone, (2) the FM microphone alone, or (3) a combination of FM and hearing instrument microphones together. 
   Usually, most of the time the FM system is used in mode (3), i.e. the FM plus hearing instrument combination (often labeled “FM+M” or “FM+ENV” mode). This operating mode allows the listener to perceive the speaker&#39;s voice from the remote microphone with a good SNR while the integrated hearing instrument microphone allows to listener to also hear environmental sounds. This allows the user/listener to hear and monitor his own voice, as well as voices of other people or environmental noise, as long as the loudness balance between the FM signal and the signal coming from the hearing instrument microphone is properly adjusted. The so-called “FM advantage” measures the relative loudness of signals when both the FM signal and the hearing instrument microphone are active at the same time. As defined by the ASHA (American Speech-Language-Hearing Association 2002), FM advantage compares the levels of the FM signal and the local microphone signal when the speaker and the user of an FM system are spaced by a distance of two meters. In this example, the voice of the speaker will travel 30 cm to the input of the FM microphone at a level of approximately 80 dB-SPL, whereas only about 65 dB-SPL will remain of this original signal after traveling the 2 m distance to the microphone in the hearing instrument. The ASHA guidelines recommend that the FM signal should have a level 10 dB higher than the level of the hearing instrument&#39;s microphone signal at the output of the user&#39;s hearing instrument. 
   When following the ASHA guidelines (or any similar recommendation), the relative gain, i.e. the ratio of the gain applied to the audio signals produced by the FM microphone and the gain applied to the audio signals produced by the hearing instrument microphone, has to be set to a fixed value in order to achieve e.g. the recommended FM advantage of 10 dB under the above-mentioned specific conditions. Accordingly,—depending on the type of hearing instrument used—the audio output of the FM receiver has been adjusted in such a way that the desired FM advantage is either fixed or programmable by a professional, so that during use of the system the FM advantage—and hence the gain ratio—is constant in the FM+M mode of the FM receiver. 
   CA 2422449 A1 relates to an example of such an FM receiver which not only receives audio signals from a remote microphone transmitter but in addition may communicate with remote devices such as a remote control or a programming unit via wireless link for data transmission. 
   EP 1 638 367 A2 relates to another example of an FM receiver for receiving audio signals from a remote microphone transmitter, wherein the FM receiver upon receipt of a polling signal from the remote microphone transmitter is capable of transmitting status information regarding the FM receiver to the remote microphone transmitter. 
   WO 97/21325 A1 relates to a hearing system comprising a remote unit with a microphone and an FM transmitter and an FM receiver connected to a hearing aid equipped with a microphone. The hearing aid can be operated in three modes, i.e. “hearing aid only”, “FM only” or “FM+M”. In the FM+M mode the maximum loudness of the hearing aid microphone audio signal is reduced by a fixed value between 1 and 10 dB below the maximum loudness of the FM microphone audio signal, for example by 4 dB. Both the FM microphone and the hearing aid microphone may be provided with an automatic gain control (AGC) unit. 
   WO 02/30153 A1 relates to a hearing system comprising an FM receiver connected to a digital hearing aid, with the FM receiver comprising a digital output interface in order to increase the flexibility in signal treatment compared to the usual audio input parallel to the hearing aid microphone, whereby the signal level can easily be individually adjusted to fit the microphone input and, if needed, different frequency characteristics can be applied. However, is not mentioned how such input adjustment can be done. 
   Contemporary digital hearing aids are capable of permanently performing a classification of the present auditory scene captured by the hearing aid microphones in order to select the hearing aid operation mode which is most appropriate for the determined present auditory scene. Examples for such hearing aids with auditory scene analyses can be found in US2002/0037087, US2002/0090098, CA 2439427 A1 and US2002/0150264. 
   Usually FM or inductive receivers are equipped with a squelch function by which the audio signal in the receiver is muted if the level of the demodulated audio signal is too low in order to avoid user&#39;s perception of excessive noise due a too low sound pressure level at the remote microphone or due to a large distance between the transmission unit and the receiver unit exceeding the reach of the FM link, see for example U.S. Pat. No. 5,734,976 and EP 1 619 926 A1. 
   As already mentioned above, usually the FM advantage is set to a value of about 10 dB, which value is a compromise taking into account a medium surrounding noise level and a good intelligibility of both the FM audio signal and the voice of the neighbours. Further, this value is based on a medium sensitivity of the hearing aid audio input and on a specific microphone impedance of the hearing aid microphone. Variations of the audio input sensitivity of different hearing aids due to microphone impedance and/or sensitivity variations will have a direct impact on the desired FM advantage of 10 dB, i.e. they will cause a deviation from this desired value, resulting in a decreasing comprehension and listening comfort. Measurements have shown audio input sensitivity variations of up to ±6 dB between the main hearing aid models present in the market. This implies that in practice the FM advantage will vary between 4 dB and 16 dB, depending on the hearing aid model connected to the FM receiver, instead of the desired value of 10 dB. In addition to that, tolerances of the FM transmitter and FM receiver gain are also added to the total FM advantage variation. Further, the desired FM advantage of 10 dB is a recommendation only and may not be optimum in any case or situation. In specific cases, the individual user&#39;s perception may require another value of the FM advantage than 10 dB. 
   It is an object of the invention to provide for a method for adjusting a system for providing hearing assistance to a user, wherein a remote microphone arrangement coupled by a wireless audio link to a receiver unit worn by the user is used and wherein perception of the transmitted audio signals should be optimized for the specific user, independently of the hearing instrument model and the FM system parameter variations and tolerances. It is a further object to provide for a corresponding system. 
   SUMMARY OF THE INVENTION 
   According to the invention, this object is achieved by a method as defined in claim  1  and by a system as defined in claim  29 , respectively. 
   The invention is beneficial in that, by transmitting test audio signals to the receiver unit, simultaneously changing the gain by transmitting corresponding gain control commands to the receiver unit until an optimum value of the gain has been determined by the user, and storing that determined optimum gain value, undesired individual deviations of the perception of the audio signals from the remote microphone arrangement from the desired condition due to individual parameter variations and individual tolerances of the system can be avoided, so that for each practical individual system the desired optimum gain applied to the audio signals of the remote microphone arrangement can be determined and stored in order to use this optimum value during normal operation of the system. 
   According to a preferred embodiment, the system comprises a hearing instrument which is worn at the user&#39;s ear and which is connected to the receiver unit or comprises the receiver unit, with the hearing instrument comprising the stimulating means, a second microphone arrangement for capturing second audio signals, and means for mixing the audio signals from the gain control unit and the second audio signals prior to stimulating the user&#39;s hearing with the mixed audio signals via said stimulating means. For such a system the individual FM advantage, i.e. the ratio of the gain applied to the audio signals from the remote microphone arrangement applied to the audio signals from the hearing instrument microphone arrangement, can by be individually optimized regardless of individual parameter variations and individual tolerances. 
   Usually the audio signals from the receiver unit and the hearing instrument microphone will be mixed in the hearing instrument in such a manner that they are processed and power-amplified together so that gain applied to these audio signals in the hearing instrument is the same for both kinds of audio signals; consequently, after mixing the gain ratio will not be changed by the usual dynamic audio signal processing of the hearing instrument. Thus, by controlling the gain applied to the audio signals from the remote microphone arrangement by the gain control unit of the receiver unit, also the gain ratio, i.e. the ratio of the gain applied to the audio signals from the remote microphone arrangement and the gain applied to the audio signals from the hearing instrument microphone, can be controlled. 
   The parameter variations and tolerances which can be compensated by the adjustment method of the present invention include the following: microphone sensitivity of the radio transmitter, modulation strength of the radio transmitter, audio output level of the radio receiver, output impedance of the radio receiver, audio input sensitivity of the hearing aid, audio input impedance of the hearing aid, and specific sensitivity of the user. 
   According to one embodiment, the test audio signals are generated by retrieving audio signals from a memory. According to another embodiment, the test audio signals may be generated by an audio signal synthesizer. According to a further alternative embodiment, the test audio signals may be generated by generating a test sound which is captured as the test audio signals by the remote microphone arrangement; usually the test sound will be the voice of a person using the transmitting unit, such as a teacher. In this case, the test sound may be captured also by the second microphone arrangement, so that for optimizing the gain, and also the gain ratio, also the audio signals captured by the second microphone arrangement may be taken into account. Typically the test audio signal transmitted to the receiver unit will be transmitted at a maximum level of the audio signals of the remote microphone arrangement, which is typical when the person using the transmitting unit is speaking. 
   A data link for transmitting the commands to the receiver unit and the audio signal link may be realized by a common transmission channel, with the bandwidth being split. 
   According to one embodiment, the system may be operated in such a manner that the gain is kept constant at a value corresponding to the determined optimum value. According to an alternative embodiment, the system may be operated in such a manner that the gain is dynamically changed according to the result of a permanently repeated auditory scene analysis based on at least one of the audio signals provided by the remote microphone arrangement and the audio signals provided by the hearing instrument microphone arrangement. In this case the determined optimum value of the gain is used to calibrate the gain control unit, i.e. the gain control algorithm is calibrated by the determined optimum gain value. 
   These and further objects, features and advantages of the present invention will become apparent from the following description when taken in connection with the accompanying drawings which, for purposes of illustration only, show several embodiments in accordance with the present invention. 

   
     BRIEF DESCRIPTION OF THE DRAWINGS 
       FIG. 1  is a schematic view of the use of an embodiment of a hearing assistance system according to the invention; 
       FIG. 2  is a schematic view of the transmission unit of the system of  FIG. 1 ; 
       FIG. 3  is a diagram showing the signal amplitude versus frequency of the common audio signal/data transmission channel of the system of  FIG. 1 ; 
       FIG. 4  is a block diagram of one embodiment of the receiver unit of the system of FIG. 
       FIG. 5  is a block diagram of one embodiment of the transmission unit of the system of  FIG. 1 ; 
       FIG. 6  is a block diagram of another embodiment of the transmission unit of the system of  FIG. 1 ; 
       FIG. 7  is a diagram showing an example of the gain set by the gain control unit versus time; 
       FIG. 8  shows schematically an example in which the receiver unit is connected to a separate audio input of a hearing aid; and 
       FIG. 9  shows schematically an example in which the receiver unit is connected in parallel to the microphone arrangement of a hearing aid. 
   

   DETAILED DESCRIPTION OF THE INVENTION 
     FIG. 1  shows schematically the use of a system for hearing assistance comprising an FM radio transmission unit  102  comprising a directional microphone arrangement  26  consisting of two omnidirectional microphones M 1  and M 2  which are spaced apart by a distance d, an FM radio receiver unit  103 , and hearing instrument  104  comprising a microphone arrangement  36 . The audio output of the receiver unit  103  is connected to an audio input of the hearing instrument  104  via an audio shoe (not shown). The transmission unit  102  is worn by a speaker  100  around his neck by a neck-loop  121  acting as an FM radio antenna, with the microphone arrangement  26  capturing the sound waves  105  carrying the speaker&#39;s voice. Audio signals and control data are sent from the transmission unit  102  via radio link  107  to the receiver unit  103  worn by a user/listener  101 . In addition to the voice  105  of the speaker  100  background/surrounding noise  106  may be present which will be both captured by the microphone arrangement  26  of the transmission unit  102  and microphone arrangement  36  of the hearing instrument  104 . Typically the speaker  100  will be a teacher and the user  101  will be a hearing-impaired person in a classroom, with background noise  106  being generated by other pupils. 
     FIG. 8  is a block diagram of an example in which the receiver unit  103  is connected to a high impedance audio input of the hearing instrument  104 . The receiver unit  103  contains a module  31  for demodulation and signal processing for processing the FM signal received by the antenna  123  from the antenna of the transmission unit  102  (these audio signals resulting from the microphone arrangement  26  of the transmission unit  102  in the following also will be referred to as “first audio signals”). The processed first audio signals are amplified by variable gain amplifier  126 . The output of the receiver unit  103  is connected to an audio input of the hearing instrument  104  which is separate from the microphone  36  of the hearing instrument  15  (such separate audio input has a high input impedance). 
   The first audio signals provided at the separate audio input of the hearing instrument  104  may undergo pre-amplification in a pre-amplifier  33 , while the audio signals produced by the microphone  36  of the hearing instrument  104  (in the following referred to “second audio signals”) may undergo pre-amplification in a pre-amplifier  37 . The hearing instrument  104  further comprises a digital central unit  35  into which the first and second audio signals are supplied as a mixed audio signal for further audio signal processing and amplification prior to being supplied to the input of the output transducer  38  of the hearing instrument  104 . The output transducer  38  serves to stimulate the user&#39;s hearing  39  according to the combined audio signals provided by the central unit  35 . 
     FIG. 9  shows a modification of the embodiment of  FIG. 8 , wherein the output of the receiver unit  103  is not provided to a separate high impedance audio input of the hearing instrument  104  but rather is provided to an audio input of the hearing instrument  104  which is connected in parallel to the hearing instrument microphone  36 . Also in this case, the first and second audio signals from the remote microphone arrangement  26  and the hearing instrument microphone  36 , respectively, are provided as a combined/mixed audio signal to the central unit  35  of the hearing instrument  104 . The gain applied to first audio signals can be adjusted by the variable gain amplifier  126  of the receiver unit  103 . Further, also the gain ratio for the first and second audio signals can be controlled by the receiver unit  103  by accordingly controlling the signal at the audio output of the receiver unit  103  and the output impedance Z 1  of the audio output of the receiver unit  103 . 
     FIG. 2  is a schematic view of the transmission unit  102  which, in addition to the microphone arrangement  26 , comprises a digital signal processor  122 , an FM transmitter  120 , an antenna  149  for establishing a short distance bidirectional inductive link  54  with an antenna  151  of the receiver unit  103 , a button  50  for activating an FM advantage adjustment mode of the transmission unit  102  and the receiver unit  103 , a button  51  to read identification information stored in the receiver unit  103  via the inductive link  54 , a button  52  for causing a “volume up” command being transmitted to the receiver unit  103 , and a button  53  for causing a “volume down” command being transmitted to the receiver unit  103 . 
   According to  FIG. 3 , the channel bandwidth of the FM radio transmitter, which, for example, may range from 100 Hz to 7 kHz, is split in two parts ranging, for example from 100 Hz to 5 kHz and from 5 kHz to 7 kHz, respectively. In this case, the lower part is used to transmit the audio signals (i.e. the first audio signals) resulting from the microphone arrangement  26 , while the upper part is used for transmitting data from the FM transmitter  120  to the receiver unit  103 . The data link established thereby can be used for transmitting control commands relating to the gain from the transmission unit  102  to the receiver  103 , and it also can be used for transmitting general information or commands to the receiver unit  103 . 
   The internal architecture of the FM transmission unit  102  is schematically shown in  FIG. 5 . As already mentioned above, the spaced apart omnidirectional microphones M 1  and M 2  of the microphone arrangement  26  capture both the speaker&#39;s voice  105  and the surrounding noise  106  and produce corresponding audio signals which are converted into digital signals by the analog-to-digital converters  109  and  110 . M 1  is the front microphone and M 2  is the rear microphone. The microphones M 1  and M 2  together associated to a beamformer algorithm form a directional microphone arrangement  26  which, according to  FIG. 1 , is placed at a relatively short distance to the mouth of the speaker  100  in order to insure a good SNR at the audio source and also to allow the use of easy to implement and fast algorithms for voice detection as will be explained in the following. The converted digital signals from the microphones M 1  and M 2  are supplied to the unit  111  which comprises a beam former implemented by a classical beam former algorithm and a 5 kHz low pass filter. The first audio signals leaving the beam former unit  111  are supplied to a gain model unit  112  which mainly consists of an automatic gain control (AGC) for avoiding an overmodulation of the transmitted audio signals. The output of a gain model unit  112  is supplied to an adder unit  113  which mixes the first audio signals, which are limited to a range of 100 Hz to 5 kHz due to the 5 kHz low pass filter in the unit  111 , and DTMF (dual-tone multi-frequency) encoded data signals supplied from a control unit  162  within a range from 5 kHz and 7 kHz. The combined audio/data signals are converted to analog by a digital-to-analog converter  119  and then are supplied to the FM transmitter  120  which uses the neck-loop  121  as an FM radio antenna. 
   The transmission unit  102  further comprises a voice memory  160  in which test audio signals are stored which can be retrieved by request of a control unit  162  and which are then supplied to the gain model unit  112 . The control unit  162  generates commands for controlling the transmission unit  102  and the receiver unit  103  according to operation of the buttons  50  to  53  by the user  100 . Such control commands are transmitted via the FM transmitter  120  and the antenna  121  to the receiver unit  103 . The units  109 ,  110 ,  111 ,  112 ,  113 ,  119  and  162  all can be realized by the digital signal processor  122  of the transmission unit  102 . 
   The receiver unit  103  is schematically shown in  FIG. 4 . The audio signals produced by the microphone arrangement  26  and processed by the units  111  and  112  of transmission unit  102  and the command signals produced by the control unit  162  of the transmission unit  102  are transmitted from the transmission unit  102  over the same FM radio channel to the receiver unit  103  where the FM radio signals are received by the antenna  123  and are demodulated in an FM radio receiver  124 . An audio signal low pass filter  125  operating at 5 kHz supplies the audio signals to a variable gain amplifier  126  from where the audio signals are supplied to the audio input of the hearing instrument  104 . The output signal of the FM radio receiver  124  is also filtered by a high pass filter  127  operating at 5 kHz in order to extract the commands from the control unit  162  contained in the FM radio signal. A filtered signal is supplied to a unit  128  including a DTMF and digital demodulator/decoder in order to decode the command signals from the control unit  162 . 
   The command signals decoded in the unit  128  are provided to a parameter update unit  129  in which the parameters of the commands are updated according to information stored in an EEPROM  130  of the receiver unit  103 . The output of the parameter update unit  129  is used to control the audio signal amplifier  126  which is gain and output impedance controlled. Thereby the audio signal output of the receiver unit  103  can be controlled according to the commands from the control unit  162  in order to control the gain (and also the gain ratio, i.e. the ratio of the gain applied to the audio signals from the microphone arrangement  26  of the transmission unit  102  and the audio signals from the hearing instrument microphone  36 ) according to the commands from the control unit  162 . 
   The inductive antenna  151  of the receiver unit  103  is connected via a unit  150  to the EEPROM  130  and is used for reading identification information stored in the EEPROM  130 , which serves to identify the receiver unit  103 , via the inductive link  54  by the transmission unit  102 . In addition, the inductive link  54  may have additional functions such as reading other receiver parameters, programming the receiver unit  103 , monitoring battery status, the receiver unit  103  and monitoring the quality of the link. 
   The desired gain determined by the amplifier  126  may be adjusted according to the following procedure. 
   First, the user  100  selects the respective receiver unit  103 , which is to be adjusted by approaching the receiver unit  103  with the transmission unit  102  so close that the receiver unit  103  comes within the reach of the inductive link  54 . Then the button  51  is pushed whereby the control unit  162  causes the transmission unit  102  to read the identification code via the inductive link  54  from the EEPROM  130  of the receiver unit  103 . Once the identification code has been read by the transmission unit  102 , this particular identification code is coded over the data link of the transmission unit  102  in order to address in the further adjustment procedure only the specified receiver unit  103 . If the user  101  uses two hearing instruments  104 , two receiver units  103  must be addressed by the transmission unit  102 . If the user  101  is the only one within the reach distance of the transmission unit  102 , the receiver identification step can be omitted. 
   As a next step, the user  100  will enter an adjustment mode of the transmission unit  102  by pushing the button  50 . 
   In the FM advantage adjustment procedure then test audio signal is generated, for example, by retrieving a test signal from the voice memory  160 . Alternatively, the test audio signals may be generated by the voice of the user  100  which is captured by the microphone arrangement  26 . In the latter case, the voice of the user  100  also will be captured by the hearing instrument microphone  36 . In any case, the test audio signal preferably will be transmitted to the receiver unit  103  at the maximum audio level of the transmission unit  102 , which is typical for the case when the user  100  is speaking. The test audio signals provided by the low pass filter  125  will be amplified by the amplifier  126  according to the presently set gain in the EEPROM  130  and then will be supplied to the hearing instrument  104  for being reproduced by the speaker  38 . 
   As a next step, perception of the test audio signals by the user  101  will be evaluated, and according to the result of this evaluation the volume-up-button  52  will be pushed if the user  101  feels that the volume of the audio test signals is too low, or the volume-down-button  53  will be pushed if the user  101  feels that the volume of the test audio signals is too high. Upon operation of the respective button  52  or  53  the control unit  162  will cause a corresponding control command to be transmitted to the receiver unit  103  where it is demodulated in the unit  128  and serves to correspondingly increase or reduce the gain applied by the amplifier  126  via the unit  129 . 
   Such change of the gain applied by the amplifier  126  is continued until an optimum value—which corresponds then to the optimum value of the individual FM advantage—has been found. Thereupon that determined optimum gain value will be stored in the EEPROM  130  of the receiver unit upon receipt of a respective command sent by the transmitting unit  102 . Such store command signal may be generated by the control unit  162  of the transmission unit  102  upon corresponding operation of the buttons at the transmission unit  102 , for example by again pushing the “A”-button  50 , or it may be generated automatically, if a certain time period without operation of the volume up or volume down-buttons  52 ,  53  has lapsed. 
   After having terminated the FM advantage adjustment procedure, the transmission unit  102  and the receiver unit  103  will resume the normal operation mode. This normal operation mode may be such that the determined optimum gain value stored in the EEPROM  130  will be continuously applied to the amplifier  126 , i.e. the amplifier  126  will be operated at constant gain. 
   According to an alternative embodiment which is shown in  FIGS. 6 and 7 , the transmission unit  102  and the receiver unit  103  may be designed such that in the normal operation mode the gain presently applied by the amplifier  126  may be changed according to the result of an auditory scene analysis permanently performed by the transmission unit  102  by analysing the audio signal captured by the microphone arrangement  26 . The receiver unit  103  shown in  FIG. 4  may be used also with the transmission unit  102  of  FIG. 6 . 
   To this end, the transmission unit  102  is provided with classification unit  134 , the functions of which may be implemented by the digital signal processor  122 . The classification unit  134  shown in  FIG. 6  includes units  114 ,  115 ,  116 ,  117  and  118 , as will be explained in detail in the following. 
   The unit  114  is a voice energy estimator unit which uses the output signal of the beam former unit  111  in order to compute the total energy contained in the voice spectrum with a fast attack time in the range of a few milliseconds, preferably not more than 10 milliseconds. By using such short attack time it is ensured that the system is able to react very fast when the speaker  11  begins to speak. The output of the voice energy estimator unit  114  is provided to a voice judgement unit  115  which decides, depending on the signal provided by the voice energy estimator  114 , whether close voice, i.e. the speaker&#39;s voice, is present at the microphone arrangement  26  or not. 
   The unit  117  is a surrounding noise level estimator unit which uses the audio signal produced by the omnidirectional rear microphone M 2  in order to estimate the surrounding noise level present at the microphone arrangement  26 . However, it can be assumed that the surrounding noise level estimated at the microphone arrangement  26  is a good indication also for the surrounding noise level present at the microphone  36  of the hearing instrument  104 , like in classrooms for example. The surrounding noise level estimator unit  117  is active only if no close voice is presently detected by the voice judgement unit  115  (in case that close voice is detected by the voice judgement unit  115 , the surrounding noise level estimator unit  117  is disabled by a corresponding signal from the voice judgment unit  115 ). A very long time constant in the range of 10 seconds is applied by the surrounding noise level estimator unit  117 . The surrounding noise level estimator unit  117  measures and analyzes the total energy contained in the whole spectrum of the audio signal of the microphone M 2  (usually the surrounding noise in a classroom is caused by the voices of other pupils in the classroom). The long time constant ensures that only the time-averaged surrounding noise is measured and analyzed, but not specific short noise events. According to the level estimated by the unit  117 , a hysteresis function and a level definition is then applied in the level definition unit  118 , and the data provided by the level definition unit  118  is supplied to the unit  116  in which the data is encoded by a digital encoder/modulator and is transmitted continuously with a digital modulation having a spectrum a range between 5 kHz and 7 kHz. That kind of modulation allows only relatively low bit rates and is well adapted for transmitting slowly varying parameters like the surrounding noise level provided by the level definition unit  118 . 
   The estimated surrounding noise level definition provided by the level definition unit  118  is also supplied to the voice judgement unit  115  in order to be used to adapt accordingly to it the threshold level for the close voice/no close voice decision made by the voice judgement unit  115  in order to maintain a good SNR for the voice detection. 
   If close voice is detected by the voice judgement unit  115 , a very fast DTMF (dual-tone multi-frequency) command is generated by a DTMF generator included in the unit  116 . The DTMF generator uses frequencies in the range of 5 kHz to 7 kHz. The benefit of such DTMF modulation is that the generation and the decoding of the commands are very fast, in the range of a few milliseconds. This feature is very important for being able to send a very fast “voice ON” command to the receiver unit  103  in order to catch the beginning of a sentence spoken by the speaker  11 . The command signals produced in the unit  116  (i.e. DTMF tones and continuous digital modulation) are provided to the adder unit  113 , as already mentioned above. 
     FIG. 7  illustrates an example of how the gain in the normal operation mode may be controlled according to the determined present auditory scene category. 
   As already explained above, the voice judgement unit  115  provides at its output for a parameter signal which may have two different values: 
   “Voice ON”: This value is provided at the output if the voice judgement unit  115  has decided that close voice is present at the microphone arrangement  26 . In this case, fast DTMF modulation occurs in the unit  116  and a control command is issued by the unit  116  and is transmitted to the amplifier  126 , according to which the gain is set to a given value which, for example, may result in an FM advantage of 10 dB under the respective conditions of for example, the ASHA guidelines. 
   “Voice OFF”: If the voice judgement unit  115  decides that no more close voice is present at the microphone arrangement  26 , a “voice OFF” command is issued by the unit  116  and is transmitted to the amplifier  126 . In this case, the parameter update unit  129  applies a “hold on time” constant  131  and then a “release time” constant  132  defined in the EEPROM  130  to the amplifier  126 . During the “hold on time” the gain set by the amplifier  126  remains at the value applied during “voice ON”. During the “release time” the gain set by the amplifier  126  is progressively reduced from the value applied during “voice ON” to a lower value corresponding to a “pause attenuation” value  133  stored in the EEPROM  130 . Hence, in case of “voice OFF” the gain of the microphone arrangement  26  is reduced relative to the gain of the hearing instrument microphone  36  compared to “voice ON”. This ensures an optimum SNR for the hearing instrument microphone  36 , since at that time no useful audio signal is present at the microphone arrangement  26  of the transmission unit  102 . 
   The control data/command issued by the surrounding noise level definition unit  118  is the “surrounding noise level” which has a value according to the detected surrounding noise level. As already mentioned above, the “surrounding noise level” is estimated only during “voice OFF” but the level values are sent continuously over the data link. Depending on the “surrounding noise level” the parameter update unit  129  controls the amplifier  126  such that according to definition stored in the EEPROM  130  the amplifier  126  applies an additional gain offset or an output impedance change to the audio output of the receiver unit  103 . 
   The application of an additional gain offset is preferred in case that there is the relatively low surrounding noise level (i.e. quiet environment), with the gain of the hearing instrument microphone  36  being kept constant. The change of the output impedance is preferred in case that there is a relatively high surrounding noise level (noisy environment), with the signals from the hearing instrument microphone  36  being attenuated by a corresponding output impedance change. In both cases, a constant SNR for the signal of the microphone arrangement  26  compared to the signal of the hearing instrument microphone  36  is ensured. 
   A preferred application of the systems according to the invention is teaching of pupils with hearing loss in a classroom. In this case the speaker  100  is the teacher, while a user  101  is one of several pupils, with the hearing instrument  104  being a hearing aid. 
   The FM advantage adjustment procedure in the adjustment mode may be similar to that described above with regard to the system of  FIGS. 4 and 5 . In the case of the embodiment of  FIGS. 6 and 7  the optimum gain value determined and stored in the adjustment mode will be used to the calibrate the gain variation based on the auditory scene analysis in the normal operation mode. In present case, for example, the value of the gain applied in the “Voice ON” regime will correspond to the optimum gain value determined and stored in the adjustment mode. 
   While in the embodiments described so far the receiver unit is separate from the hearing instrument, in some embodiments it may be integrated with the hearing instrument. 
   The microphone arrangement producing the second audio signals may be connected to or integrated within the hearing instrument. The second audio signals may undergo an automatic gain control prior to being mixed with the first audio signals. The microphone arrangement producing the second audio signals may be designed as a directional microphone comprising two spaced apart microphones. 
   While various embodiments in accordance with the present invention have been shown and described, it is understood that the invention is not limited thereto, and is susceptible to numerous changes and modifications as known to those skilled in the art. Therefore, this invention is not limited to the details shown and described herein, and includes all such changes and modifications as encompassed by the scope of the appended claims.