Patent Publication Number: US-6993480-B1

Title: Voice intelligibility enhancement system

Description:
BACKGROUND OF THE INVENTION 
     1. Field of the Invention 
     The present invention relates to intelligible reproduction of human speech or voice sounds, and more particularly, relates to systems for improving the intelligibility of voice sounds or signals that are degraded in some fashion, such as degradation caused by noise. 
     2. Description of the Related Art 
     Speech reproduction systems, such as public address systems, telephones, cellular telephones, two-way radios, broadcast radios, etc., are often used in environments where the listener hears the speech signal combined with noise. In some circumstances the noise is of such a level that intelligibility of the desired spoken communication from the speech reproduction system is greatly degraded. 
     A typical speech reproduction system includes a signal source that generates a speech signal, a loudspeaker, and a transmission system that carries the speech signal from the source to the loudspeaker. Typical signal sources include microphone, tape playback units, audio units, computer speech generators, etc. The types of noise in a typical speech reproduction system can be loosely categorized into three general groups based on the point where the noise enters the system, the noise groups include: source noise, transmission noise, and ambient noise. Source noise is noise introduced at the source. Wind noise in a microphone is an example of source noise. Transmission noise is noise introduced by the transmission system, that is, noise introduced between the source and the loudspeaker. A common example of transmission noise is the static that is sometimes heard in a telephone, cellular telephone, or radio broadcast. Ambient noise is noise present in the listener&#39;s environment, that is, acoustic noise that the listener hears in addition to the sounds from the loudspeaker. For example, the background noise heard in a noisy environment such as an airport or automobile is ambient noise. 
     There are many environments of this type where communication is lost, or at least partly lost, because the ambient noise level masks or distorts the speaker&#39;s voice, as it is heard by the listener. These environments include airports, subway, bus and railroad terminals, aircraft and trains, aircraft carriers, landing craft, helicopters, dock facilities, cars and other vehicles, and other noisy places. Few people who have attempted to understand a public announcement or use a telephone in a noisy airport can fail to appreciate the difficulty of extracting useful information in the presence of such ambient noise. 
     Attempts to minimize loss of intelligibility in the presence of noise have involved use of equalizers, clipping circuits, or simply increasing the volume of the sound from the loudspeaker system. Equalizers and clipping circuits may themselves increase the overall noise level, and thus fail to solve the problem. Simply increasing the overall level of sound from the loudspeaker does not significantly improve intelligibility and often causes other problems such as feedback and listener discomfort. 
     SUMMARY OF THE INVENTION 
     The present invention solves these and other problems by providing improved intelligibility of voice communication that would otherwise be degraded by noise. In one embodiment, intelligibility of speech is improved by a speech enhancer that uses an aural filter in combination with a speech expander. The speech enhancer also improves the intelligibility of speech that is degraded by factors other than noise, such as, for example, speech that is mumbled. 
     The speech enhancer provides a transfer function that approximates the inverse (or compliment) of the Fletcher-Munson (F-M) curves. The F-M curves quantify the way in which the human hearing system, particularly the ear, processes sounds. As demonstrated by the F-M curves, the frequency response of the human hearing system is non-linear. The human hearing system favors the middle frequency sounds over low frequency and high frequency sounds. When the sounds are relatively quiet (e.g., low volume levels) the hearing system strongly favors middle frequency sounds. As the sound increases in volume, the frequency response of the hearing system becomes flatter (e.g., more uniform) and the middle frequency sounds are not favored as much. 
     The input signal to the speech enhancer is typically a speech signal, such as, for example, the signal from a microphone, tape deck, CD player, etc. When the speech signal is operating at a low volume level, the speech enhancer provides a transfer function that is relatively flatter than the transfer function at high volume levels. For example, when an announcer speaking into the microphone is talking very quietly, more of the low and high frequency components of the announcer&#39;s voice are provided to the listener. This provides the listener with more information in order to help the listener understand the words. Conversely, when the speech signal is operating at high volume levels, the speech enhancer provides a transfer function that produces relatively more gain in the middle frequency ranges than in the low and high frequency ranges. Intelligibility of the speech is enhanced because it is the middle frequencies that contribute most to the intelligibility of speech. At higher volume levels, the lower and higher frequencies merely contribute to the overall sound volume level and thus tend to increase listener discomfort and feedback rather than intelligibility. 
     Stated differently, the speech enhancer provides a transfer function that is in many respects, complementary to the transfer function of the human hearing system. By providing a complementary transfer function, the speech enhancer improves intelligibility, and listener comfort, by reducing the relative volume level of sounds that do not contribute to (or even reduce) speech intelligibility. The speech enhancer may advantageously be used in or in connection with: public address systems; hearing aids; communication devices, including telephones and cellular telephones; audio processors for improving clarity and/or intelligibility of music, speech or the spoken word; apparatus for use in processing audio electronic signals consisting primarily of speech to improve intelligibility and/or clarity; integrated circuits; video monitors; video tuners; stereo receivers and amplifiers; tape decks; car stereos; televisions; portable stereos; boomboxes; stereo processors for use in cinemas; video disc playback and/or recording apparatus; audio playback and/or recording apparatus; home audio-visual recording apparatus; laser disc players and records; VCRs; digital versatile disk (DVD) players; digital video tape players; speakers; speaker systems containing a sound transducer and an integral amplifier; CD (compact disc) playback and/or recording devices; motion picture projectors; cable television receivers and decoders; remote control units for these goods; computer programs having sound generating capability; computer software for expanding an audio image generated by speakers for use in the entertainment field; computers; computer sound processing cards; industry standard computer interface cards; computer audio processing circuitry; computer hardware, namely computer diskettes, computer floppy disks, hard discs, CD-ROM discs, digital video discs, optical storage discs, and computer solid-state cartridges; audio and/or audio-visual recordings stored on magnetic tape or optical media; audio and/or audio-visual prerecorded media containing entertainment material in the form of the spoken word, music and other sounds, namely motion picture film, VCR cassette tapes, laser discs, video discs, optical discs analog or digital audio cassette tapes, and analog or digital video cassette tapes; and the like. 
     One embodiment provides for enhancing the intelligibility of voice information, such as spoken words, recorded speech, synthesized speech, and the like, projected into an area of ambient noise from a loudspeaker system that receives an input signal derived from an electrical voice signal representing spoken words. The electrical voice signal may come from a microphone, a playback device, a receiver, etc. For convenience, the voice signal is described herein as an electrical signal with the understanding that the electrical voice signal may also be embodied as a sequence of digital values, as in a computer or digital signal processor. The electrical signal is provided to an aural filter that provides relatively less attenuation of middle (e.g., speech) frequencies of the electrical signal and relatively more attenuation of other frequencies. The filtered signal is provided to a voice expander having a varying gain. 
     The gain of the expander is varied according to some property of the filtered signal. For example, the gain of the expander may be varied according to the envelope of the filtered signal, the average power in the filtered signal, the average Root Mean Square (RMS) value of the filtered signal, the average peak value of the filtered signal, etc. An output of the voice expander is combined with the electrical voice signal to produce an enhanced voice signal. The enhanced voice signal is amplified and may then be provided to one or more loudspeakers to be projected as sound into an area of ambient noise. Alternatively, the enhanced voice signal may be provided to a recording device and recorded for later playback. The enhanced voice signal may also be provided to a loudspeaker in a communications device, such as, for example, a telephone, cellular telephone, cordless telephone, radio, or other communications receiver. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
       The advantages and features of the disclosed invention will readily be appreciated by persons skilled in the art from the following detailed description when read in conjunction with the drawings listed below. 
         FIG. 1A  is a block diagram of a system that includes speech enhancement. 
         FIG. 1B  is a block diagram of an audio system, such as a cellular telephone system, that provides enhanced speech from a transmission or recording medium. 
         FIG. 1C  is a block diagram of an audio system, such as a public address system, that provides enhanced speech from a loudspeaker system. 
         FIG. 2  is a frequency-domain plot of the spectrum response of typical human speech. 
         FIG. 3  is a frequency-domain plot of the Fletcher-Munson equal loudness contours for tones in a frontal sound field for humans of average hearing acuity. 
         FIG. 4  is a signal processing block diagram of a speech enhancer having an aural filter and a speech expander. 
         FIG. 5  is a frequency-domain plot of one embodiment of an aural filter combined with a speech expander. 
         FIG. 6  is a time-domain plot showing the time-amplitude response of one embodiment of a voice expander circuit. 
         FIG. 7  is a frequency-domain plot of a typical speech vocalization showing a modulated carrier and a modulation envelope. 
         FIG. 8A  is a frequency-domain plot showing amplitude response curves for the speech enhancer shown in  FIG. 4 . 
         FIG. 8B  is a frequency-domain plot showing the improvement provided by the speech enhancer of  FIG. 4  as compared to a system that merely increases the volume of speech sounds. 
         FIG. 9A  is a block diagram, with frequency domain plots, showing the operation of the system of  FIG. 4  for relatively low volume sounds when the noise source is upstream of the speech enhancer. 
         FIG. 9B  is a block diagram, with frequency domain plots, showing the operation of the system of  FIG. 4  for relatively high volume sounds when the noise source is upstream of the speech enhancer. 
         FIG. 9C  is a block diagram, with frequency domain plots, showing the operation of the system of  FIG. 4  for relatively low volume sounds when the noise source is downstream of the speech enhancer. 
         FIG. 9D  is a block diagram, with frequency domain plots, showing the operation of the system of  FIG. 4  for relatively high volume sounds when the noise source is downstream of the speech enhancer. 
         FIG. 10  shows one embodiment of a circuit diagram that implements the speech enhancer shown in  FIG. 4 . 
         FIG. 11  is a circuit diagram of one implementation of an aural filter. 
         FIG. 12  is a block diagram of one embodiment of a speech expander. 
         FIG. 13  is a circuit diagram of one implementation of the speech expander shown in  FIG. 12 . 
     
    
    
     In the drawings, the first digit of any three-digit number generally indicates the number of the figure in which the element first appears. Where four-digit reference numbers are used, the first two digits indicate the figure number. 
     DETAILED DESCRIPTION 
       FIG. 1A  illustrates a generic system having a speech enhancer  106 . Speech signals are provided by a speech source  103 . The speech source  103  is any device that provides a speech signal, such as an analog signal or a digital data stream. The speech source  103  includes, for example, a person talking into a microphone or a speech generating device such as a computer speech program. An output of the speech source  103  is provided to an input of an optional signal processing block  105 . An output of the signal processing block  105  is provided to an input of the speech enhancer  106 . An output of the speech enhancer  106  is provided to an input of an optional signal processing block  113 . An output of the optional signal processing block  113  is provided to a loudspeaker  112 . 
     The optional signal processing blocks  105  and  113  represent the signal processing and transmission operations normally performed on the speech signal as the signal travels from the source  103  to the loudspeaker  112 . Typical operations performed in the optional signal processing bocks  105  and/or  113  may include, for example, filtering, amplification, gain control, feedback cancellation, mixing, transmission, storage, playback, reception, encoding, decoding, noise canceling, up-conversion, down-conversion, detection, modulation, etc. The loudspeaker  112  is any device that converts the speech signal into an acoustic signal, including, for example, a cone-type loudspeaker, a horn-type loudspeaker, an earphone, a headset, a telephone handset loudspeaker, a speakerphone loudspeaker, an impedance transformer, etc. 
       FIG. 1B  is a block diagram that illustrates the speech enhancer  106  in a communication system or a recording/playback system. Communication systems include, for example, telephones, cellular telephones, cordless telephones, satellite systems (including the IRIDIUM system), spread-spectrum radios, two-way radios, walkie-talkies, marine radios, HAM radios, aircraft radios, broadcast radios, shortwave radios, Citizen&#39;s Band (CB) radios, dispatch radios (e.g., for taxicab and truck drivers), police radios, military communications systems including VHF, frequency-hopping, and spread-spectrum systems, intercom systems, video-conferencing systems, optical networks, and computer networks (including the Internet). 
     In  FIG. 1B , the source  103  comprises a person (announcer)  102  speaking into a microphone  104 . The microphone  104  may be located, for example, in a telephone, cellular telephone, cordless telephone, cockpit voice recorder, radio, tape recorder, computer, etc. In  FIG. 1B , the microphone is shown located in a cellular or cordless telephone handset  127  comprising the microphone  104  and a transceiver (transmitter/receiver) that includes a sender such as a transmitting system  107 . The transmitting system  107  sends information over a communication channel. The transmitting system  107  comprises an optional speech enhancer  106 , an optional audio processing block  108  and a transmitting device  109 . The output of the microphone  104  is provided to the speech enhancer  106  and the output of the speech enhancer  106  is provided to an input of an optional audio-processing block  108 . The output of the optional audio-processing block  108  is provided to an input of a transmitter (or recording) device  109 . 
     An output from the transmitting device  109  is provided to an input of a repeater  129  (e.g., a cellular telephone tower, a base station, a satellite, etc.). An output of the repeater  129  is provided to an input of a receiving (or playback) device  111 . An output of the receiving device  111  is provided to the input of an optional speech enhancer  106 . An output of the speech enhancer  106  is provided to an input of an amplifier  110  and an output of the amplifier  110  is provided to the loudspeaker  112 . The receiving device  111 , speech enhancer  106 , and the amplifier  110  are shown as elements of a transceiver that includes a receiving system  130  located in a telephone handset  131 . An optional user control  132  is provided to allow the user  114  to control the operation of the speech enhancer  106 . The control  132  may include, for example, a switch, a button, a thumb control, a menu item, etc. In some embodiments, the control  132  is used to enable and disable the speech enhancer  106 . In some embodiments, the control  132  is used to control the amount of enhancement provided by the speech enhancer  106 . 
     The speech enhancer  106  is interposed anywhere in the signal path between the microphone  104  and the loudspeaker  112 . Thus, for example, the speech enhancer  106  may be provided in the transmitter system  107  as shown, in the base station  129  as shown, or in the receiver system  130  as shown. 
     The transmitting/recording device  109  may be a radio transmitter (e.g., a microwave transmitter in a telephone or cellular telephone system), optical transmitter, fiber-optic transmitter, acoustic transmitter etc., that converts the voice signals into signals that propagate in a transmission medium to the receiving device  111 . The repeater  129  is typical of many communications system. However, is some applications, such as, for example, walkie-talkies or other two-way radios, the repeater  129  is sometimes omitted. 
     Alternatively, the transmitting/recording device  109  may be a recording device configured to record on a storage media, and the receiving/playback device  111  is configured to retrieve data from the storage media. Typical storage media includes magnetic tape, optical disks, computer disks, film, compact disks, magneto-optical disks, solid-state memories, bubble memories, etc. 
       FIG. 1C  illustrates the basic components of a typical public address system having a speech enhancer  106 .  FIG. 1C  shows the source  103  comprising the announcer  102  speaking into the microphone  104 . The microphone  104  converts the speech sounds into electrical speech signals and provides the electrical speech signals to the speech enhancer  106 . One skilled in the art will recognize that one or more amplifiers, often called pre-amplifiers, may be provided between the output of the microphone  104  and the input of the speech enhancer  106  in order to amplify the weak electrical signals provided by the microphone  104 . An output of the speech enhancer  106  is provided to an input of the optional audio-processing block  108 . The processing block  108  may provide, for example, feedback suppression, long distance distribution systems such as line-transformers or repeaters, etc. An output of the processing block  108  is provided to an input of the amplifier  110 . The optional audio-processing block  108  may also be omitted, in which case, the output of the speech enhancer  106  is provided directly to the input of the amplifier  110 . An output of the amplifier  110  is provided to the loudspeaker  112 . 
     The speech enhancer  106  modifies the electrical signals provided by the microphone  104  such that the voice sounds projected by the loudspeaker system  112  have enhanced intelligibility, even in the presence of noise. The loudspeaker may be located to project sound in a listener area to be heard by one or more listeners. The listener area may be, for example, a home, an office (e.g., from an office PA system or a speaker-phone), an auditorium, an airplane cabin, an airport, a stadium, a shopping center, a fairground, etc. 
     In one embodiment, the speech enhancer  106  takes advantage of the manner in which human speech is generated, heard, and processed by the individual human ear and brain. The speech enhancer  106  enhances vocal sounds, including, for example, formants of vowels, consonants, fricatives and plosives according to the way in which the human ear hears and perceives speech sounds, such that the enhanced vocal sounds provide a speech signal of increased intelligibility. 
     A brief description of mechanics of speech generation and comprehension will help to explain some aspects of the present invention. Human speech is produced by generating sounds in the vocal tract. The vocal tract causes these sounds to resonate at different frequencies. Vowels are generated by an air stream expelled from the lungs to cause vibration of the human vocal folds, generally known as vocal cords. Sound generated by vibration of the vocal cords is composed of a fundamental frequency or base band and many harmonic partials or overtones, at successively higher frequencies. Amplitudes of the harmonics decrease with increasing frequency at a rate of about 12 decibels per octave. The baseband, or fundamental frequency, and its overtones pass through the vocal tract, which includes various cavities within the throat, head and mouth that provide a plurality of individual resonances. The vocal tract has a plurality of characteristic modes of resonance and to some extent acts as a plurality of resonators operating on the base band or fundamental frequency and its overtones. Because of the selective resonating action of the vocal tract, amplitudes of the several partials of the fundamental frequency of the vocal cords do not decrease in a smooth curve with increasing frequency, but exhibit sharp peaks at frequencies corresponding to the particular resonances of the vocal tract. These peaks or resonances are termed “formants”. 
       FIG. 2  is a frequency-domain graph of a voiced sound (e.g. a vowel), plotting amplitude against frequency of a number of harmonics. At the left side of the graph, at the lowest frequency, is the fundamental frequency or base band caused by vibration of the vocal cords. This base band frequency is typically between about 60 and 250 hertz for a typical adult male voice. The many harmonics of the fundamental frequency are indicated by the individual components, such as the components  201 ,  202 , and  203  shown in  FIG. 2 . It can be seen that the entire voice signal is made up of the base band and a large number of individual harmonics over the entire frequency band. The frequency band of interest in voice signals is generally between about 60 and about 7,500 Hz (Hertz). 
       FIG. 2  illustrates the fact that the individual harmonics, which have amplitudes that naturally decrease with increasing frequency, do not decrease in amplitude in a smooth curve, but rather exhibit certain peaks, such as those indicated at  206 ,  208 , and  210 . These peaks represent the individual resonances of the vocal tract and are illustrated for purposes of exposition as being three in number, although there may be as many as four, five or more in an ordinary human vocal tract. These peaks, or vocal tract resonances, are the formants of the spoken voice. In an adult male the first four (lower frequency) formants are typically close to about 500, 1500, 2500 and 3500 hertz, respectively. 
     Moving the various articulatory organs (including the jaw, the body of the tongue, the tip of the tongue) changes frequency of the several formants over a wide range. Different formant frequencies have different sensitivities to shape or position of individual articulatory organs. It is the selected movement of these organs that each human speaker employs to give voice to a selected speech sound. Conversely, when listening to spoken words each speech sound can be recognized, in part, by its set of formants. 
     Normal human speech includes voiced sounds and unvoiced sounds. Voiced sounds are those caused by vibration of the vocal cords in the air stream generated by the lungs and comprise the vowels of the spoken word. Unvoiced sounds are those that are generated by the vocal tract in the absence of vibration of the vocal cords. The discussion given above with respect to voiced sounds and the formants of  FIG. 2  is also applicable to unvoiced sounds, which also have formants caused by resonant cavities of the vocal tract. Unvoiced sounds include consonants, plosives and fricatives. These sounds are generated by action of the tongue, teeth and mouth, which control the release of air from the lungs, but without vibration of the vocal cords. These include sounds of various consonants. Unvoiced sounds include sounds of spoken words involving the letters M, N, L, Z, G (as in frigid), DG (as in judge), etc. These plosives, fricatives, and consonants, although not involving vocal cord vibration, nevertheless have characteristic frequencies, generally higher than the fundamental frequency of vocal cord vibration, and often in the range of 2,000 to 3,000 hertz. Regardless of whether sound produced in the vocal tract is generated by vibration of the vocal cords (voiced sounds), or is generated without vibration of the vocal cords (consonants, plosives, and fricatives), the vocal tract resonances typically operate to produce formants which are resonant peaks in different ones of the harmonics of the generated fundamental frequency. 
     It has been found that the formants in the human speech make a significant contribution to intelligibility of speech to the listener. That is, the human listener will recognize specific vowels or consonants, plosives, or fricatives by the particular pattern of its formants. This is the pattern of relative frequencies of the several formants. The formant pattern may be based upon fundamental frequencies of higher or lower pitch, such as the higher pitch of the voice of a woman or a child, or the lower pitch of the voice of a man. The pattern of formants, being the relative frequencies of resonant peaks, identifies to the listener the nature of the spoken sound. 
     There are two components to intelligibility of speech. The first component is speech generation, as discussed above. The second component is speech hearing and perception, or, in other words, the way in which the human hearing system receives and processes speech sounds. The human hearing system is known to be nonlinear. Moreover, the frequency response of the human hearing is dependent on the loudness, or volume, of the sounds being heard.  FIG. 3  shows equal loudness contours, often referred to as the Fletcher-Munson curves, for tones in a frontal sound field for humans of average hearing acuity. The loudness level in phons corresponds to the sound pressure levels at 1000 Hz, where, by definition, a 1-kHz tone of a 20 dB sound pressure level has a loudness level of 20 phons. 
     The contours shown in  FIG. 3  can be viewed as inverted frequency response curves of the ear for different sound pressure levels. To give the same sensation of the 20 phon loudness at 100 Hz as 1 kHz, the sound pressure level must be increased about 17 dB. To give the 20 phon loudness at 20 Hz requires a sound pressure level about 62 dB higher than at 1 kHz. This means that the sensitivity of the ear is much less at lower frequencies than at 1 kHz. From the contours in  FIG. 3 , it is evident that the frequency response of the human ear is, in general, similar to a bandpass-type response which is flatter at higher sound pressure levels. 
     Different frequencies contained in the spoken voice contribute different amounts to intelligibility of the spoken word. Mid-band frequencies, in the order of about 1.5 to 3.5 kHz, contribute relatively larger percentages to intelligibility. For example, broken down by octaves in the frequency range of about 250 hertz to 5 Kilohertz and above, the octave centered at 250 hertz contributes approximately 7.2% to intelligibility of the spoken voice heard by a human listener, the octave centered at 500 hertz contributes approximately 14.4%, and that centered at 1 kilohertz contributes approximately 22.2%. The octave centered at 2 kilohertz contributes approximately 32.8%, and the octave centered at 4 kilohertz contributes approximately 23.4%. 
     Table 1 below indicates percentage contribution to intelligibility of different frequency components of a human voice signal that is broken down into one-third octave frequency bands or full octave frequency bands. 
     
       
         
           
               
               
               
             
               
                 TABLE 1 
               
               
                   
               
               
                   
                 % Contribution 
                 % Contribution 
               
               
                 Band Center Frequency Hz 
                 One-Third Octave 
                 Octave 
               
               
                   
               
             
            
               
                   
               
            
           
           
               
               
               
               
            
               
                 200 
                 and below 
                 1.2 
                   
               
               
                 250 
                   
                 3.0 
                 7.2 
               
               
                 315 
                   
                 3.0 
               
               
                 400 
                   
                 4.2 
               
               
                 500 
                   
                 4.2 
                 14.4 
               
               
                 680 
                   
                 6.0 
               
               
                 800 
                   
                 6.0 
               
               
                 1 
                 kHz 
                 7.2 
                 22.2 
               
               
                 1.25 
                 kHz 
                 9.0 
               
               
                 1.6 
                 kHz 
                 11.2 
               
               
                 2 
                 kHz 
                 11.4 
                 32.8 
               
               
                 2.5 
                 kHz 
                 10.2 
               
               
                 3.15 
                 kHz 
                 10.2 
               
               
                 4 
                 kHz 
                 7.2 
                 23.4 
               
               
                 5 
                 kHz and above 
                 6.0 
               
               
                   
               
            
           
         
       
     
     One embodiment of the present invention uses the manner in which speech is generated, and the manner in which speech is heard, to provide speech intelligibility enhancement. The various voiced and unvoiced sounds are filtered and selectively amplified to enhance intelligibility, even in the presence of noise. According to embodiments disclosed herein, voice intelligibility is enhanced by selectively filtering and expanding the components of a speech signal according to the way in which the human hearing system processes speech sounds. 
       FIG. 4  is a signal processing block diagram  400  of one embodiment of the speech enhancer  106  shown in  FIG. 1 . The speech enhancer  400  uses an aural filter  406  to provide spectral shaping of the speech signal and a speech expander  408  to generate a time-dependent enhancement factor.  FIG. 4  may also be used as a flowchart to describe a program running on a DSP or other processor which implements the signal processing operations of an embodiment of the present invention. 
       FIG. 4  shows an input  402  and an output  404 . The input  402  is provided to a first input of the aural filter  406 , and to a first input of a combiner  410 . An output of the aural filter  406  is provided to an input of the speech expander  408 . An output of the speech expander  408  is provided to second input of the combiner  410 . An output of the combiner  410  is provided to the output  404 . 
       FIG. 4  is illustrative to show one signal processing embodiment of the present invention. As such,  FIG. 4  is, in some respects, an illustration of a mathematical formula that describes the manipulations performed on the voice signal. One skilled in the art will recognize that, as with most mathematical formulas, the sequence of signal processing operations shown in  FIG. 4  can be combined, separated, factored, and otherwise manipulated without changing the transfer function of the block diagram  400 . Thus, for example, the feedforward path from the input  402  to the second input of the combiner  410  need not be shown explicitly. The feedforward path can be merged into the aural filter  406  and the speech expander  408 . The feedforward path has been made explicit in  FIG. 4  for the purpose of clarity of description, and not as a limitation. 
     In an alternative embodiment, the input  402  is also provided to a gain control input of the speech expander  408  such that the gain of the speech expander is controlled, by at least a portion of the input voice signal. 
     The speech enhancer provides a transfer function that approximates the inverse (or compliment) of the familiar Fletcher-Munson (F-M) curves shown in  FIG. 3 . The F-M curves quantify the way in which the human hearing system, particularly the ear, process sounds. As demonstrated by the F-M curves, the frequency response of the human hearing system is non-linear. The human hearing system favors middle frequency sounds over low frequency and high frequency sounds. When the sounds are relatively quiet (e.g., low volume levels) the hearing system strongly favors middle frequency sounds. As the sound increases in volume, the frequency response of the hearing system becomes flatter and the middle frequency sounds are not favored as much. 
     The input signal to the speech enhancer is a speech signal. When the speech signal is operating at a low volume level, the speech enhancer provides a transfer function that is relatively flatter than the transfer function at high volume levels. Conversely, when the speech signal is operating at high volume levels, the speech enhancer provides a transfer function that produces relatively more gain in the middle frequency ranges than in the low and high frequency ranges. Thus, for example, when an announcer speaking into the microphone is talking very quietly, more of the low and high frequency components of the announcer&#39;s voice are provided to the listener. This provides the listener with more information in order to help the listener understand the words. 
     For a fixed volume setting (such as the volume setting in a public address system) the speech enhancer compensates for the volume of an announcer&#39;s voice. For example, when the announcer speaks loudly into the microphone, relatively fewer of the low and high frequency components are provided to the listener. This provides the listener with relatively less information (frequency content) but less information is sufficient because the announcer is talking loudly. The additional information in the low and high frequencies would only serve to increase the overall volume level without adding significantly to the intelligibility of the words. Moreover, when the speaker talks loudly, and the sounds get louder, the hearing system of the listener is more able to perceive the low and high frequency sounds. Thus, even though at high volume levels the speech enhancer is attenuating the low and high frequency sounds with respect to the middle frequency sounds, the listener will not necessarily perceive the full extent of the relative attenuation because the listener&#39;s hearing system is providing relatively less attenuation of the low and high frequency sounds. 
     Stated differently, the speech enhancer is a dynamic filter that provides a transfer function that is a function of one or more properties of the input signal. In one embodiment, the transfer function of the dynamic filter is a function of the volume level of the voice signal (like the human ear wherein the transfer function is a function of the sound pressure level). In one embodiment, the transfer function of the speech enhancer is, in some respects, approximately complementary to the transfer function of the human hearing system. By providing a complementary transfer function, the speech enhancer improves intelligibility, and listener comfort, by reducing the relative volume level of: sounds that are irritating; sounds that do not contribute to (or even reduce) speech intelligibility; sounds that the human hearing system is more able to perceive; and sounds that might cause annoying feedback. 
       FIG. 5  is a frequency-domain plot that shows a family of six curves that illustrate the general shape of the combined transfer function of the aural filters  406  and speech expander  408 . The family of six curves shows a generally bandpass characteristic with a transmission peak in the 2 kHz to 3 kHz range. A curve  502  shows the transfer function of the aural filter  406  alone (i.e., when the speech expander  408  is configured to provide a transfer function of unity). In one embodiment, the speech expander is an amplifier whose gain is a function of the input signal. Thus, as the input signal increases in amplitude, the gain of the speech expander also increases in amplitude. The increase in gain is given by an expansion factor e. In one embodiment, the gain g of the speech expander may be express by the relationship g=k(1+ei), where k is a constant and i is related to the amplitude of the input signal. As discussed below, i may related to the envelope of the input signal, the time average power of the input signal, the Root-Mean-Square (RMS) average of the input signal, etc. When the expansion factor e is zero, then the gain of the speech expander is unity (for k=1), corresponding to the curve  502 . 
       FIG. 5  also shows curves  504 ,  506 ,  508 ,  510  and  512  corresponding approximately to e=0.2, 0.4, 0.6, 0.8, and 1.0 respectively. The amplitude dependence of the gain can be seen by comparing the curve  502  with the curve  512 . The curve  502  corresponds to the input of the speech expander (and thus also the output of the speech expander for e=1). At 200 Hz, the amplitude of the curve  502  is approximately −16 dB and the amplitude of the curve  512  at the output of the speech expander is approximately −7 dB, corresponding to a gain of 9 dB. By contrast, at 2000 Hz, the amplitude of the curve  502  is approximately −1 dB and the amplitude of the curve  512  is approximately 16 dB, corresponding to a gain of 17 dB. The curves shown in  FIG. 5  are approximately the inverse of the F-M curves shown in  FIG. 3  in the range of about 100 Hz to about 20 kHz. 
     In one embodiment, the speech expander  408  uses an Automatic Gain Control (AGC) comprising a linear amplifier with an internal servo feedback loop. The servo automatically adjusts the average amplitude of the output signal to match the average amplitude of a signal at the control input. The average amplitude of the control input is typically obtained by detecting the envelope of the control signal. The control signal may also be obtained by other methods, including, for example, lowpass filtering, bandpass filtering, peak detection, RMS averaging, mean value averaging, etc. 
     In the speech expander, portions of the input signal are provided to the control input. In response to an increase in the amplitude of the envelope of the signal provided to the input of the speech expander  408 , the servo loop increases the forward gain of the speech expander  408 . Conversely, in response to a decrease in the amplitude of the envelope of the signal provided to the input of the speech expander  408 , the servo loop decreases the forward gain of the speech expander  408 . In one embodiment, the gain of the speech expander  408  increases more rapidly that the gain decreases.  FIG. 6  is a time domain plot that illustrates the gain of the speech expander  408  in response to an input tone burst having an envelope that is a unit step. One skilled in the art will recognize that  FIG. 6  is a plot of gain as a function of time, rather than an output signal as a function of time. Most amplifiers have a gain that is fixed, however, the automatic gain control (AGC) in the speech expander  408  varies the gain of the speech expander  408  in response to some characteristic (such as the envelope) of the input signal. 
     The envelope unit step input is plotted as a curve  605  and the gain is plotted as a curve  602 . In response to the leading edge of the envelope pulse  605 , the gain rises during a period  604  corresponding to an attack time constant period  604 . At the end of the time period  604 , the gain  605  reaches a steady-state gain of A 0 . In response to the trailing edge of the envelope pulse  605  the gain falls back to zero during a period  606  corresponding to a decay time constant period  606 . The attack time constant period  604  and the decay time constant period  606  are desirably selected to provide enhancement of the speech signal while reducing listener discomfort and feedback. 
     An understanding of the action of the speech expander can be shown in connection with a speech waveform shown in a plot  700  in  FIG. 7A . The plot  700  shows a higher-frequency portion  704  that is amplitude modulated by a lower-frequency portion having a modulation envelope  706 . The higher frequency portion  704  corresponds to the formants and other tones produced by the vocal cords. The modulation envelope  706  corresponds to the modulation of the formants and other sounds produced by moving the articulatory organs. Since the vocal chords typically vibrate much faster than the movement of the other articulatory organs, the sound produced by the vocal chords is modulated in amplitude, and frequency, by the other body parts. Short fast speech sounds, such as the consonants in western speech will typically have a modulation envelope that is relatively short with a fast risetime and a high (loud) peak. A vowel sound, on the other hand, will typically have a modulation envelope that is relatively long with a slow risetime and a low peak. 
       FIG. 8A  shows a frequency-domain plot of the amplitude response of the speech enhancer  400 . The frequency selection provided by the aural filter  406  biases the action of the speech expander  408  towards a speech (middle) frequency region primarily between about 1 kHz and 5 kHz. In the lower frequency region, the speech enhancer  400  provides a transfer function that approaches unity. In the higher frequency region, the speech enhancer  400  provides relatively less gain than in the speech frequency region. 
     In the speech region, the speech enhancer  400  provides a varying transfer function, owing to the variable gain of the speech expander  408 .  FIG. 8A  shows a family of gain curves in the speech frequency region, corresponding to input signals with different envelope amplitudes. A curve  802  shows the gain of the speech enhancer  400  for speech signals with a relatively low amplitude. The curve  802  is approximately uniform at 0 dB, showing a slight rise to approximately 4 dB in the middle frequency region. A curve  808  shows the gain of the speech enhancer  400  for speech signals with a relatively large amplitude. The curve  808  rises from approximately 0 dB at low frequencies to almost 20 dB at the middle frequencies and falls below 10 dB at high frequencies. A comparison of the curve  802  with the curve  808  shows that for input signals with a relatively higher envelope amplitude, the gain of speech enhancer  400  in the speech frequency region is larger than the gain for signal with a relatively lower envelope amplitude. 
     The speech enhancer  400  advantageously shapes the spectrum of the speech signal according to the amplitude of the signal.  FIG. 8B  show some aspects of the difference between the speech enhancer  400  and a simple volume control.  FIG. 8B  shows the curve  808 , corresponding to relatively high volume signals.  FIG. 8B  also shows a curve  810 , which is the curve  802  (from  FIG. 8A ) simply increased by a uniform gain of approximately 15 dB. Thus, the curve  810  corresponds to the action of a simple volume control on the curve  802 . A hatched region between the curves  810  and  808  represents extra sound energy that would be heard by the listener  114 . In other words, the hatched region represents sound that is suppressed by the speech enhancer circuit  400  at relatively high volume levels. This same sound would not be suppressed by a conventional speech system. The extra sound represented by the hatched region is less important for intelligibility, but rather, merely increases the overall sound level, and possible discomfort, perceived by the listener  114 . By suppressing sounds in the hatched region, the speech enhancer advantageously improves intelligibility while reducing the overall sound output level, and thereby, increasing listener comfort. 
     The speech enhancer  400  improves intelligibility of voice sounds in the presence of noise, regardless of whether the source of the noise is upstream (before) the speech enhancer or downstream (after) the speech enhancer.  FIG. 9A  shows the operation of the speech enhancer  106  in a system operating at relatively low volume levels where the source of the noise is upstream of the speech enhancer  106 . In  FIG. 9A , an output of a speech source  902  is provided to a first input of an adder  912 . An output of a noise source  904  is provided to a second input of the adder  912 . An output of the adder  912  is provided to the input of the speech enhancer  106 . An output of the speech enhancer  106  is provided to a process block  908 . The process block  908  represents the response of the human ear (i.e., the ear of the listener  114 ). An output of the process block  908  is provided to a speech perception block  910 . The speech perception block  910  represents the speech perception of the listener  114 . 
     A frequency-domain plot  901  shows an example of a frequency response plot of the output from the speech source  902 . A frequency-domain plot  903  shows another exemplary frequency response plot of the output from the noise source  904 . A frequency-domain plot  905  shows an exemplary frequency response plot of the output from the speech adder  912 . A frequency-domain plot  907  shows an exemplary frequency response plot of the output from the speech enhancer  106 . A frequency-domain plot  909  shows an exemplary frequency response plot of the output from the process block  908 . 
     As shown in the plot  901 , most of the frequency components of the speech signal from the source  902  lie in a middle frequency range having a bandwidth B. As shown in the plot  905 , when the amplitude of the speech signal is relatively low, then the noise will contaminate the speech. For speech signals of relatively low amplitude, the gain of the speech enhancer  106  is relatively uniform, and thus the plot  907  is similar to the plot  905 . However, at low volume levels, the human ear is relatively more sensitive to sounds within the bandwidth B and relatively less sensitive to sounds outside the bandwidth B. Thus, the plot  909  shows that more of the information within the bandwidth B reaches the speech perception block  910 . The relatively uniform response curve of the speech enhancer  106  at low volume levels means that a substantial portion of the available speech is signal is provided to the listener  114 , thus providing the listener  114  with more information. 
       FIG. 9B  is similar to  FIG. 9A , however,  FIG. 9B  shows the operation of the speech enhancer  106  in a system operating at relatively high volume levels. A frequency-domain plot  921  shows an exemplary frequency response plot of the output from the speech source  902 . A frequency-domain plot  923  shows an exemplary frequency response plot of the output from the noise source  904 . A frequency-domain plot  925  shows an exemplary frequency response plot of the output from the adder  912 . A frequency-domain plot  927  shows an exemplary frequency response plot of the output from the speech enhancer  106 . A frequency-domain plot  929  shows an exemplary frequency response plot of the output from the process block  908 . 
     For speech signals of relatively high amplitude, the gain of the speech enhancer  106  is higher in the middle frequency regions than in the low and high frequency regions, and thus the plot  927  has a high frequency rolloff and a low frequency rolloff not seen in the plot  905 . The rolloff at high and low frequencies reduces the low and high frequency components of the noise without significantly reducing the portions of the signal containing speech information. At high volume levels, the response of the human ear is relatively uniform, and thus, the plot  929  is similar to the plot  927 . 
       FIG. 9C  shows the operation of the speech enhancer  106  in a system operating at relatively low volume levels where the source of the noise is downstream of the speech enhancer  106 . In  FIG. 9C , the output of the speech source  902  is provided to the input of the speech enhancer  106 . The output of the speech enhancer  106  is provided to the first input of the adder  912 . The output of the noise source  904  is provided to the second input of the adder  912 . The output of the adder  912  is provided to the input the process block  908 . The output of the process block  908  is provided to the speech perception block  910 . 
     A frequency-domain plot  941  shows an exemplary frequency response plot of the output from the speech source  902 . A frequency-domain plot  943  shows an exemplary frequency response plot of the output from the noise source  904 . A frequency-domain plot  945  shows an exemplary frequency response plot of the output from the speech enhancer  106 . A frequency-domain plot  947  shows an exemplary frequency response plot of the output from the adder  912 . A frequency-domain plot  909  shows an exemplary frequency response plot of the output from the process block  908 . 
       FIG. 9C  shows that for speech signals of relatively low amplitude, the gain of the speech enhancer  106  is relatively uniform, and thus the plot  945  is similar to the plot  941 . The speech enhancer  106  does not significantly reduce the amplitude of the low or high frequency components of the speech signal. The relatively uniform response curve of the speech enhancer  106  at low volume levels means that a substantial portion of the available speech is signal is provided at the output of the speech enhancer  106  so that the noise signal is less likely to degrade the speech signal (especially the low and high frequency components of the speech signal). 
       FIG. 9D  is similar to  FIG. 9C , however,  FIG. 9D  shows the operation of the speech enhancer  106  in a system operating at relatively high volume levels. A frequency-domain plot  961  shows an exemplary frequency response plot of the output from the speech source  902 . A frequency-domain plot  963  shows an exemplary frequency response plot of the output from the noise source  904 . A frequency-domain plot  965  shows an exemplary frequency response plot of the output from the speech enhancer  106 . A frequency-domain plot  967  shows an exemplary frequency response plot of the output from the adder  912 . A frequency-domain plot  969  shows an exemplary frequency response plot of the output from the process block  908 . 
     For speech signals of relatively high amplitude, the gain of the speech enhancer  106  is significantly higher in the bandwidth B than in the low and high frequency regions outside B. Thus, the plot  965  has a low frequency rolloff and a high frequency rolloff not seen in the plot  961 . The rolloff at low and high frequencies reduces the low and high frequency components of the speech signal that are relatively less important for intelligibility, thus minimizing the potential for listener discomfort at high volume levels. At high amplitudes, the noise signal  963  is less likely to degrade the voice signal  965 , and thus the plot  967  is similar to the plot  965  inside the bandwidth B. At high volume levels the frequency response of the human ear, as represented by the process block  908 , is relatively uniform and thus the signal  969  is similar to the signal  967 . 
       FIG. 10  is a circuit schematic showing one embodiment of the speech enhancer  400  shown in  FIG. 4 . In  FIG. 10 , an input  1002  is provided to a first terminal of a DC-blocking capacitor  1003  and to a first terminal of a DC-blocking capacitor  1006 . The input  1002  is provided voice information from a voice source, such as the source  103 , including, for example, a microphone, a transducer, a speech generator, a receiver, a computer, etc. 
     A second terminal of the capacitor  1003  and a second terminal of the capacitor  1006  are provided to a first terminal of a resistor  1008 . The first terminal of the resistor  1008  is also provided to a non-inverting input of an operational amplifier (op-amp)  1010 . A second terminal of the resistor  108  is provided to ground. 
     An output of the op-amp  1010  is provided to an inverting input of the op-amp  1010 , to an input of an aural filter  1012 , and to a first terminal of a resistor  1020 . An output of the aural filter  1012  is provided to an input of a speech expander  1014 . An output of the speech expander  1014  is provided to a first fixed terminal of a potentiometer  1016 . A second fixed terminal of the potentiometer  1016  is provided to ground and a wiper of the potentiometer  1016  is provided to a first throw of a single pole double throw (SPDT) switch  1018 . The second throw of the SPDT switch  1018  is provided to ground. The pole of the SPDT switch  1018  is provided to a first terminal of a resistor  1026 . 
     Returning to the resistor  1020 , a second terminal of the resistor  1020  is provided to an inverting input of an op-amp  1024  and to a first terminal of a resistor  1022 . A non-inverting input of the op-amp  1024  is provided to ground. An output of the op-amp  1024  is provided to a second terminal of the resistor  1022  and to a first terminal of a resistor  1028 . 
     A second terminal of the resistor  1026 , and a second terminal of the resistor  1028  are provided to an inverting input of an op-amp  1032 . A non-inverting input of the op-amp  1032  is provided to ground. An output of the op-amp  1032  is provided to a first terminal of a feedback resistor  1030 . A second terminal of the feedback resistor  1030  is provided to the inverting input of the op-amp  1032 . The output of the op-amp  1032  is also provided to a first terminal of a DC-blocking capacitor  1036  and to a first terminal of a DC-blocking capacitor  1038 . 
     A second terminal of the capacitor  1036  and a second terminal of the capacitor  1038  are provided to a first terminal of a resistor  1040 . The first terminal of the resistor  1040  is provided to an output  1004  and a second terminal of the resistor  1040  is provided to ground. 
     The resistors  1026 ,  1028 , and  1030  in combination with the op-amp  1032  are shown as a combiner  1034 . 
     In one embodiment, the DC-blocking capacitors  1003  and  1036  are 4.7 uF capacitors and the capacitors  1006  and  1038  are 0.01 uF capacitors. The resistor  1008  is a 100 k-ohm resistor, the resistor  1040  is a 2.7 k-ohm resistor, and the resistors  1028 ,  1030 , and  1032  are 10 k-ohm resistors. The potentiometer is a 1.0 k-ohm linear potentiometer. The op-amps  1010 ,  1024 , and  1032  are TL074 op-amps supplied by Texas Instruments, Inc. (or any other similar amplifiers). 
     The output of the speech expander  1014  is an enhanced speech signal that is combined with the speech input signal (provided at the output of the op-amp  1024 ) by the combiner  1034 . The optional switch  1018  is provided to disable the speech enhancement processing by disconnecting the signal path from the speech expander  1014  to the combiner  1034 . The potentiometer  1016  is provided to allow an adjustment of the amount of speech enhancement by selecting the amount of enhanced speech signal that is provided to the combiner  1034 . 
     The potentiometer  1016  controls the amount of speech enhancement. An enhanced signal is provided at the output of the speech expander  1014 . The enhanced signal is added to the input signal from the input  1002  by the combiner  1034 . The potentiometer controls how much of the enhanced signal is combined with the input signal to produce an output signal at the output  1004 . The potentiometer  1016  controls the amount of enhanced signal that is combined with the input signal to produce the output signal. The switch  1016  is provided to disable the speech enhancement processing such that the output signal at the output  1004  is linearly similar to the input signal at the input  1002 . 
     One embodiment of the aural filter  1012  is shown in  FIG. 11 , where the aural filter  1012  has an input  1102  and an output  1104 . The input  1102  is provided to a first terminal of a resistor  1106 , to a first terminal of a resistor  1118 , and to a first terminal of a resistor  1130 . A second terminal of the resistor  1106  is provided to a first terminal of a resistor  1110  and to a first terminal of a capacitor  1108 . A second terminal of the resistor  1110  is provided to a first terminal of a resistor  1112  and to a first terminal of a resistor  1114 . A second terminal of the resistor  1114  is provided to a second terminal of the capacitor  1108  and to a first terminal of a resistor  1116 . A second terminal of the resistor  1116  is provided to an output of an op-amp  1140 . 
     Returning to the resistor  1118 , a second terminal of the resistor  1118  is provided to a first terminal of a resistor  1122  and to a first terminal of a capacitor  1120 . A second terminal of the resistor  1122  is provided to a first terminal of a resistor  1126  and to a first terminal of a capacitor  1124 . A second terminal of the resistor  1126  is provided to a second terminal of the capacitor  1120  and to a first terminal of a resistor  1128 . A second terminal of the resistor  1128  is provided to an output of the op-amp  1140 . 
     A second terminal of the resistor  1112  and a second terminal of the capacitor  1124  are provided to an inverting input of the op-amp  1140 . 
     Returning to the resistor  1130 , a second terminal of the resistor  1130  is provided to a first terminal of a capacitor  1134  and to a first terminal of a resistor  1132 . A second terminal of the resistor  1132  is provided to the output of the op-amp  1140 . A second terminal of the capacitor  1134  is provided to a first terminal of a capacitor  1136  and to a first terminal of a resistor  1138 . A second terminal of the resistor  1138  is provided to ground, and a second terminal of the capacitor  1136  is provide to the inverting input of the op-amp  1140 . 
     A non-inverting input of the op-amp  1140  is provided to ground, and the output of the op-amp  1140  is provided to the output  1104 . 
     In a preferred embodiment, the op-amp  1140  is a TL074 op-amp, and the values for the resistors and capacitors in the aural filter  1012  are listed in Table 2 below. 
     
       
         
           
               
               
               
               
               
             
               
                   
                 TABLE 2 
               
               
                   
                   
               
               
                   
                   
                 Resistance 
                   
                 Capacitance 
               
               
                   
                 Resistor 
                 (k-ohms) 
                 Capacitor 
                 (uF) 
               
               
                   
                   
               
             
            
               
                   
               
            
           
           
               
               
               
               
               
            
               
                   
                 1106 
                 11.0 
                 1108 
                 0.047 
               
               
                   
                 1110 
                 84.5 
                 1120 
                 0.0022 
               
               
                   
                 1112 
                 11.0 
                 1124 
                 0.01 
               
               
                   
                 1114 
                 10.7 
                 1134 
                 0.0047 
               
               
                   
                 1116 
                 11.0 
                 1136 
                 0.1 
               
               
                   
                 1118 
                 3.65 
               
               
                   
                 1122 
                 6.34 
               
               
                   
                 1126 
                 97.6 
               
               
                   
                 1128 
                 3.65 
               
               
                   
                 1130 
                 0.95 
               
               
                   
                 1132 
                 453.0 
               
               
                   
                 1138 
                 0.274 
               
               
                   
                   
               
            
           
         
       
     
     A block diagram of one embodiment of the speech expander  1014  is shown in  FIG. 12  as a block diagram, and a corresponding circuit diagram is shown in  FIG. 13 . In  FIG. 12 , an input  1203  is provided to a first input of a fixed gain amplifier  1206 , to a first input of a variable gain amplifier  1208 , and to a first terminal of a resistor  1205 . A second terminal of the resistor  1205  is provided to a first terminal of a grounded resistor  1207  and to an input of an envelope detector  1212 . An output of the envelope detector  1212  is provided to an attack/decay buffer  1210 . An output of the attack/decay buffer  1210  is provided to a gain control input of the gain-controlled amplifier  1208 . An output of the fixed gain amplifier  1206  is provided to a first input of an output adder  1207  and an output of the variable gain amplifier  1208  is provided to a second input of the output adder  1207 . An output of the output adder  1207  is provided to a speech expander output  1204 . 
     The fixed gain amplifier  1206  provides a unity gain feedforward path to the output adder  1204 . Thus, even if the gain of the gain-controlled amplifier  1208  is zero, the feedforward path will provide the speech expander  1014  with a minimum gain of 1.0. The resistors  1205  and  1207  are connected as a voltage divider to select a portion of the input signal provided at the input  1203 . The selected portion is provided to the envelope detector  1212 . The output of the envelope detector is a signal that approximates the envelope of the input signal. The envelope signal is provided to the attack/decay buffer. When the envelope signal has a positive slope (rising edge) the attack/decay buffer provides a signal to increase the gain of the gain-controlled amplifier at a rate given by the attack time constant. When the envelope signal has a negative slope (falling edge) the attack/decay buffer provides a signal to decrease the gain of the gain-controlled amplifier at a rate given by the decay time constant. 
     The speech expander  1014  shown in  FIG. 12  is an expander because the gain of the speech expander  1014 , and thus the output level, is controlled by the input signal. As the average amplitude of the envelope of the input signal increased, the gain increases. Conversely, as the average amplitude of the envelope of the input signal level decreases, the gain decreases. The voltage divider (resistors  1205  and  1207 ) is desirably constructed to provide sufficient expansion of the input signal to enhance the intelligibility of speech. 
       FIG. 13  is a circuit diagram illustrating one embodiment of the speech expander  1014 . In  FIG. 13 , the input  1203  is provided to a first terminal of a capacitor  1342  and to the first terminal of the resistor  1205 . The second terminal of the resistor  1205  is provided to a first terminal of a capacitor  1306  and to the first terminal of the grounded resistor  1207 . A second terminal of the capacitor  1306  is provided to a first terminal of a resistor  1308  and a second terminal of the resistor  1308  is provided to an envelope detector input (pin  3 ) of a gain control circuit  1349 . In one embodiment, the gain control circuit  1349  is an NE572. 
     The NE572 is a dual-channel, high-performance gain control circuit in which either channel may be used for dynamic range compression or expansion. Each channel has a full-wave rectifier to detect the average value of input signal, a linearized, temperature-compensated variable gain cell and a dynamic time constant buffer. The buffer permits independent control of dynamic attack and recovery time with minimum external components and improved low-frequency gain control ripple distortion. Pin-outs for the NE572 are listed in Table 3 (where n,m designates channels A,B). The NE572 is used in the present embodiments as an inexpensive, low-noise, low distortion, gain controlled amplifier. One skilled in the art will recognize that other gain-controlled amplifiers can be used as well. 
     
       
         
           
               
               
               
             
               
                   
                 TABLE 3 
               
               
                   
                   
               
               
                   
                 Pin 
                 Function 
               
               
                   
                   
               
             
            
               
                   
                 1,15 
                 Tracking Trim 
               
               
                   
                 2,14 
                 Recovery 
               
               
                   
                 3,13 
                 Rectifier input 
               
               
                   
                 4,12 
                 Attack 
               
               
                   
                 5,11 
                 Vout 
               
               
                   
                 6,10 
                 THD trim 
               
               
                   
                 7,9  
                 Vin 
               
               
                   
                  8 
                 Ground 
               
               
                   
                 16 
                 Vcc 
               
               
                   
                   
               
            
           
         
       
     
     A first terminal of an attack timing capacitor  1343  is provided to an attack control input (pin  4 ) of the gain control circuit  1349  and a second terminal of the attack timing capacitor  1343  is provided to ground. A first terminal of a decay timing capacitor  1344  is provided to a decay control input (pin  2 ) of the gain control circuit  1349  and a second terminal of the decay timing capacitor  1344  is provided to ground. 
     A second terminal of the capacitor  1342  is provided to a V in  terminal (pin  7 ) of the gain control circuit  1349  and to a first terminal of a resistor  1310 . A second terminal of the resistor  1310  is provided to a V out , terminal (pin  5 ) of the gain control circuit  1349  and to an inverting input of an op-amp  1347 . A non-inverting input of the op-amp  1347  is provided to a terminal of a grounded capacitor  1346 , to a non-inverting input of an op-amp  1352 , and to a first terminal of a resistor  1345 . A second terminal of the resistor  1345  is provided to a THD terminal (pin  6 ) of the gain control circuit  1349 . 
     An output of the op-amp  1347  is provided to the output  1204  and to a first terminal of a feedback resistor  1349 . A second terminal of the feedback resistor  1349  is provided to the inverting input of the op-amp  1347 . 
     An inverting input of the op-amp  1352  is provided to a terminal of a grounded resistor  1343  and to a first terminal of a feedback resistor  1351 . A second terminal of the feedback resistor  1351  is provided to an output of the op-amp  1352  and to a first terminal of a resistor  1350 . A second terminal of the resistor  1350  is provided to the inverting input of the op-amp  1347 . 
     In one embodiment, the capacitors  1342 ,  1306 , and  1346  are 2.2 uF capacitors. The attack timing  1343  capacitor is a 0.10 uF capacitor and the decay timing capacitor  1344  is a 1.0 uF capacitor. The resistor  1348  is a 3.1 k-ohm resistor, and the resistors  1345  is a 1.0 k-ohm resistor. The resistors  1353  and  1351  are 10 k-ohm resistors, and the resistors  1310 ,  1349 , and  1350  are 17.4 k-ohm resistors. 
     The gain control circuit  1349  includes an envelope detector  1361 , an attack/decay buffer  1362 , and a gain element  1363 . As in the block diagram in  FIG. 12 , an output of the envelope detector  1361  is provided to the attack/decay buffer  1362 , and an output of the attack/decay buffer  1362  controls the gain element  1363 . The attack and delay time constants are controlled by resistor-capacitor (RC) networks. The attack/decay buffer  1362  provides an internal 10 k-ohm resistor for the attack RC network and an internal 10 k-ohm resistor for the decay RC network. The 0.1 uF attack capacitor  1343  produces an attack time constant of approximately 4.0 ms (milliseconds). The 1.0 uF decay capacitor  1344  produces a decay time constant of approximately 40.0 ms. In other embodiments the attack time constant may range from 1 ms to 40 ms and the decay time constant may range from 10 ms to 100 ms. 
     The gain element  1363  is similar to an electronically variable resistor and used in connection with the feedback circuit of the op-amp  1347  to vary the gain of the op-amp  1347 . The op-amp  1352  provides a DC bias. The unity gain feedforward path is provided by the resistor  1310 . 
     Recordings 
     As described above,  FIG. 1B  illustrates use of voice processing methods and apparatus of the present invention applied to a voice communication system. It will be readily appreciated that the same voice processing can be applied to the making of any suitable recording, which is later employed as the sound input to a conventional playback system. In making such a recording, using the voice processing and intelligibility enhancement techniques described herein, the resulting recording inherently includes the intelligibility enhancement provided by the processing circuitry. Therefore, no further intelligibility enhancement processing is needed when such a recording is played through a conventional playback system. 
     To make such a recording there is used a system substantially the same as that shown in  FIG. 1B , so that the sound recorded on the tape or other record medium includes the enhanced speech signal processed by the system  400  shown in  FIG. 4 . 
     The described processing will also provide an intelligibility enhanced recording where the input sound comprises a spoken voice that originates in a noisy environment. Such a condition exists in many situations, such as, for example, in the case of a cockpit voice recorder (CVR), which is a recording device carried in the cockpit of commercial aircraft for the purpose of making a record of occurrences and conversations of the personnel in the aircraft cockpit. The cockpit environment is exceedingly noisy, so that, in the past, recordings made by the cockpit voice recorder have been difficult to comprehend because of their degraded intelligibility. 
     The present invention is applicable to such a cockpit voice recorder to enhance intelligibility of the recorded sound when played back on conventional playback equipment. An intelligibility enhanced cockpit voice recorder of the present invention is substantially the same as the system illustrated in  FIG. 1B . 
     OTHER EMBODIMENTS 
     Although the foregoing has been a description and illustration of specific embodiments of the invention, various modifications and changes can be made thereto by persons skilled in the art, without departing from the scope and spirit of the invention as defined by the following claims.