Patent Publication Number: US-7916876-B1

Title: System and method for reconstructing high frequency components in upsampled audio signals using modulation and aliasing techniques

Description:
REFERENCE TO PROVISIONAL PATENT APPLICATION 
     This patent application claims priority to U.S. Provisional Patent Application No. 60/483,750 that was filed on Jun. 30, 2003. 
    
    
     TECHNICAL FIELD OF THE INVENTION 
     The present invention is generally directed to digital communications technology and, in particular, to a system and method for providing artificial signal enhancement to digital audio signals in digital communication devices. 
     BACKGROUND OF THE INVENTION 
     There is a demand for techniques to improve the quality of telephone audio. The quality of telephone audio is also referred to as voice quality or audio quality. The major limitation in increasing audio quality is the size of the telephone spectrum bandwidth. The telephone spectrum bandwidth ranges from three hundred Hertz (300 Hz) to three thousand four hundred Hertz (3,400 Hz). Because the transmission bandwidth is fixed in size, it is not possible to transmit better quality audio signals. 
     Instead, a technique must be developed that artificially improves the audio quality after the audio signal has been received in a handset but before the audio signal is sent through the speaker of the handset. That is, the enhancement of the audio signal can take place only in a receiving handset. 
     There is a need in the art for a system and method for creating an audio signal that has an enhanced audio quality. There is a need in the art for a system and method for enhancing an audio signal after the audio signal has been received in a receiver of a handset but before the audio signal is sent to a speaker of the handset. 
     SUMMARY OF THE INVENTION 
     To address the above-discussed deficiencies of the prior art, it is a primary object of the present invention to provide a system and method for enhancing the voice quality of a digital audio signal in a receiving handset. 
     The present invention enhances a digital audio signal in a receiving handset by upsampling the audio signal to expand the audio bandwidth of the received audio signal. Then one or more additional signals are added to the audio signal. The additional signals have higher frequencies than the original audio signal. The addition of the higher frequency additional signals is done using harmonic modulation and aliasing techniques. 
     In particular, the present invention improves speech quality by extending the telephone frequency band (three hundred Hertz (300 Hz) to three thousand four hundred Hertz (3,400 Hz)) to a frequency band of ten Hertz (10 Hz) to eight thousand Hertz (8,000 Hz). This is done by the controlled addition of audio signals in frequency bands that are normally filtered out and therefore not sent. The frequency range of the frequency bands that are normally filtered out is from approximately three thousand four hundred Hertz (3,400 Hz) to approximately ten thousand Hertz (10,000 Hz). 
     The system and method of the present invention reconstructs the high frequency components of the digital audio signal using a harmonic enhancer in a baseband integrated circuit of the receiver handset. The original spectrum of the digital audio signal is upsampled in a times N upsample unit (where N is greater than or equal to two (2)) to double the size of the bandwidth. A low pass filter then removes a high frequency alias of the original spectrum. The spectrum is then modulated with a first carrier frequency and sent to a first filter bank where a low pass filter and a high pass filter shape the modulated harmonic spectrum. After gain adjustment, the modulated harmonic spectrum is added to a delayed version of the original spectrum. Additional harmonic spectra are similarly created at other carrier frequencies and added to the audio output spectra to reconstruct high frequency components of the digital audio signal. 
     It is an object of the present invention to provide a system and method for enhancing the voice quality of a digital audio signal in a receiving handset. 
     It is also an object of the present invention to provide a system and method for enhancing a digital audio signal after the digital audio signal has been received in a receiver of a handset but before the digital audio signal is sent to a speaker of the handset. 
     It is yet another object of the present invention to provide a system and method for upsampling a digital audio signal to expand the audio bandwidth of the audio signal. 
     It is still another object of the present invention to provide a system and method for creating one or more harmonic spectra and adding the harmonic spectra to the audio output spectra of the digital audio signal to reconstruct high frequency components of the digital audio signal. 
     The foregoing has outlined rather broadly the features and technical advantages of the present invention so that those skilled in the art may better understand the detailed description of the invention that follows. Additional features and advantages of the invention will be described hereinafter that form the subject of the claims of the invention. Those skilled in the art should appreciate that they may readily use the conception and the specific embodiment disclosed as a basis for modifying or designing other structures for carrying out the same purposes of the present invention. Those skilled in the art should also realize that such equivalent constructions do not depart from the spirit and scope of the invention in its broadest form. 
     Before undertaking the Detailed Description of the Invention below, it may be advantageous to set forth definitions of certain words and phrases used throughout this patent document: the terms “include” and “comprise,” as well as derivatives thereof, mean inclusion without limitation; the term “or,” is inclusive, meaning and/or; the phrases “associated with” and “associated therewith,” as well as derivatives thereof, may mean to include, be included within, interconnect with, contain, be contained within, connect to or with, couple to or with, be communicable with, cooperate with, interleave, juxtapose, be proximate to, be bound to or with, have, have a property of, or the like; and the term “controller” means any device, system or part thereof that controls at least one operation, such a device may be implemented in hardware, firmware or software, or some combination of at least two of the same. It should be noted that the functionality associated with any particular controller may be centralized or distributed, whether locally or remotely. Definitions for certain words and phrases are provided throughout this patent document, those of ordinary skill in the art should understand that in many, if not most instances, such definitions apply to prior uses, as well as future uses, of such defined words and phrases. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
       For a more complete understanding of the present invention and its advantages, reference is now made to the following description taken in conjunction with the accompanying drawings, in which like reference numerals represent like parts: 
         FIG. 1  illustrates a prior art baseband integrated circuit that is capable of converting an analog audio signal to a digital audio signal for transmission to a remotely located receiver; 
         FIG. 2  illustrates a baseband integrated circuit that is capable of receiving a transmitted digital audio signal and converting the digital audio signal to an analog audio signal using harmonic enhancement techniques in accordance with the principles of the present invention; 
         FIG. 3  illustrates a block diagram of an advantageous embodiment of a harmonic enhancer of the present invention; 
         FIG. 4  illustrates an exemplary original digital audio spectrum for input into a times two (2) upsample unit of the harmonic enhancer of the present invention; 
         FIG. 5  illustrates an exemplary digital audio spectrum that represents the output of the times two (2) upsample unit of the harmonic enhancer of the present invention for the input spectrum shown in  FIG. 4  showing an alias of the original input spectrum; 
         FIG. 6  illustrates the effect of a low pass filter on the exemplary output spectrum shown in  FIG. 5  to illustrate how the alias of the spectrum may be filtered away by the low pass filter to leave the original input spectrum and a double size bandwidth; 
         FIG. 6A  illustrates the result of low pass filtering the spectrum shown in  FIG. 6 ; 
         FIG. 7  illustrates the effect of modulating the original input spectrum at an exemplary modulation frequency of two kilohertz; 
         FIG. 8  illustrates the effect of a low pass filter of a filter block on the spectrum shown in  FIG. 7  to illustrate how the higher frequencies of the spectrum may be filtered away by the low pass filter of the filter block; 
         FIG. 9  illustrates the effect of a high pass filter of a filter block on the spectrum shown in  FIG. 8  to illustrate how the lower frequencies of the spectrum may be filtered away by the high pass filter of the filter block; 
         FIG. 10  illustrates the effect of applying both a low pass filter and a high pass filter of a filter block to the modulated spectrum shown in  FIG. 7 ; 
         FIG. 11  illustrates the addition of an attenuated version of the spectrum shown in  FIG. 10  to the original input spectrum shown in  FIG. 4 ; 
         FIG. 12  illustrates the addition of two additional similarly modulated spectra to the spectrum shown in  FIG. 11 ; 
         FIG. 13  illustrates a graph of signal magnitude (in decibels) versus frequency (in kiloHertz) for the audio spectrum of the audio signal showing the effect of applying harmonic enhancement to the original audio signal; 
         FIG. 14  illustrates a flow chart showing the steps of a first portion of an advantageous embodiment of the method of the present invention; and 
         FIG. 15  illustrates a flow chart showing the steps of a second portion of an advantageous embodiment of the method of the present invention. 
     
    
    
     DETAILED DESCRIPTION OF THE INVENTION 
       FIGS. 1 through 15 , discussed below, and the various embodiments used to describe the principles of the present invention in this patent document are by way of illustration only and should not be construed in any way to limit the scope of the invention. Those skilled in the art will understand that the principles of the present invention may be implemented in any type of suitably arranged digital audio system. 
       FIG. 1  illustrates a prior art circuit  100  that is capable of converting an analog audio signal to a digital audio signal and wirelessly transmitting the digital audio signal to a remotely located receiver. Prior art circuit  100  generally comprises a microphone  110 , a baseband integrated circuit  120 , a transmitter  170 , and an antenna  180 . The baseband integrated circuit  120  comprises a codec  130 , a sampler  140 , a generic digital signal processor  150 , and an Adaptive Differential Pulse Code Modulation (ADPCM) coder  160 . An example of baseband integrated circuit  120  is the SC14428 baseband integrated circuit chip manufactured by National Semiconductor Corporation. 
     An analog audio signal from microphone  110  is received by codec  130 . Codec  130  is responsible for the initial audio filtering. Codec  130  filters the analog audio signal from three hundred Hertz (300 Hz) up to and including three thousand four hundred Hertz (3400 Hz). Then the filtered signal is sent to sampler  140 . Sampler  140  then samples the signal on eight kilohertz (8 kHz). The digitized signal then proceeds to the generic digital signal processor (GenDSP)  150 . Several different signal processes may be carried out with the digital audio signal in GenDSP  150  before the digital audio signal is sent to the ADPCM coder  160 . 
     After the digital audio signal processing in GenDSP  150  has been completed, the digital audio signal is sent to the ADPCM coder  160 . ADPCM coder  160  codes the digital audio signal and sends it to transmitter  170  for transmission through antenna  180 . 
       FIG. 2  illustrates a receiving system  200  that comprises a baseband integrated circuit  230  that is capable of receiving the transmitted digital audio signal and converting the digital audio signal to an analog audio signal using harmonic enhancement techniques in accordance with the principles of the present invention. Baseband integrated circuit  230  comprises an ADPCM decoder  240 , a generic digital signal processor (GenDSP)  250 , a harmonic enhancer  260 , and a codec  270 . 
     The digital audio signal from antenna  180  of transmitter  170  of  FIG. 1  is received through antenna  210  of receiver  220  of  FIG. 2 . The received signal is first decoded in the ADPCM decoder  240 . The decoded signal is then sent to GenDSP  250  for further signal processing. GenDSP  250  comprises harmonic enhancer  260  of the present invention. The structure and operation of harmonic enhancer  260  will be described more fully below. 
     After the digital audio signal has been processed by GenDSP  250 , the digital audio signal is sent to codec  270  to be filtered and converted to an analog signal. Codec  270  has a sample frequency of sixteen kiloHertz (16 kHz) and a bandwidth of eight kiloHertz (8 kHz). Then the resulting analog audio signal is sent to speaker  280  and broadcast through speaker  280 . 
       FIG. 3  illustrates a block diagram of an advantageous embodiment of harmonic enhancer  260  of the present invention. Harmonic enhancer  260  comprises a times two (2) upsample unit  310 , low pass filter  315 , delay unit  320 , adder  325 , modulators ( 330 ,  345  and  360 ), a first filter block  335 , a second filter block  350 , a third filter block  365 , and gain units ( 340 ,  355 ,  370 ). 
     The function of the harmonic enhancer  260  is to “add” additional sound to the original audio spectrum in order to improve the quality of voice communications. In the advantageous embodiment of the harmonic enhancer  260  shown in  FIG. 3  there are three branches. The first branch comprises modulator  330 , first filter block  335  and gain unit  340 . The second branch comprises modulator  345 , second filter block  350  and gain unit  355 . The third branch comprises modulator  360 , third filter block  365  and gain unit  370 . It is understood that the use of three branches is an example and that the harmonic enhancer of the present invention is not limited to exactly three branches. Specifically, the harmonic enhancer may have more than three branches or fewer than three branches. As will be more fully explained, the three branches use their respective carrier frequencies to create the additional sounds that are to be added to the audio spectrum. 
     The original audio spectrum has a sample rate of eight kilohertz (8 kHz). This means that after four kilohertz (4 kHz) no audio will be present. To be more precise, the audio spectrum runs from three hundred Hertz (300 Hz) to three thousand four hundred Hertz (3400 Hz). To obtain a bandwidth of eight kilohertz (8 kHz) the bandwidth must be doubled. The bandwidth may be doubled by upsampling. 
     For example, consider the exemplary original digital audio spectrum  410  shown in  FIG. 4  as the input to harmonic enhancer  260 .  FIG. 4  illustrates a graph  400  of the magnitude of the input signal  410  versus frequency. The input to the harmonic enhancer  260  is normally the signal that would be supplied to codec  270  if the harmonic enhancer  260  were not present within GenDSP  250 . The input signal  410  is first provided to the times two (2) upsample unit  310 . The times two (2) upsample unit  310  doubles the bandwidth of the audio spectrum by upsampling. The structure and operation of the times two (2) upsample unit  310  is well known in the art and will not be described here. 
       FIG. 5  illustrates an exemplary digital audio spectrum that represents the output of the times two (2) upsample unit of the harmonic enhancer  260  for the input spectrum  410  shown in  FIG. 4 . The upsampling operation causes an alias of the original spectrum around the Nyquist frequency (four kilohertz (4 kHz)) as shown in  FIG. 5 . That is,  FIG. 5  illustrates a graph  500  of the magnitude of the input signal  410  versus frequency and the alias  510  of the input signal versus frequency. As seen in  FIG. 5 , the bandwidth has been doubled up to eight kilohertz (8 kHz). 
     The digital audio signal is then passed out through low pass filter  315 . Low pass filter  315  filters out the unwanted alias  510  of the spectrum because the alias  510  has a mirrored spectrum. The graph  600  of  FIG. 6  illustrates the filter characteristic  610  of the low pass filter  315 . The filter characteristic  610  of low pass filter  315  preserves the low frequency portion  410  of the audio spectrum and filters out the high frequency alias  510  of the audio spectrum. Removing the high frequency alias  510  leaves the original audio spectrum  410  in a bandwidth that has now doubled to eight kiloHertz (8 kHz). The resulting spectrum is shown in  FIG. 6A . 
     Then the output of low pass filter  315  is provided to the input of delay unit  320 , to the input of modulator  330 , to the input of modulator  345 , and to input of modulator  360 . The delay unit  320  delays the original spectrum  410  by a fixed time in order to compensate for the group delay of the filter blocks in the three branches to create the additional audio spectra to be added back to the original spectrum  410 . 
     Consider the operation of the first branch that comprises modulator  330 , first filter block  335  and gain unit  340 . Modulator  330  modulates the original spectrum  410  with a first carrier frequency (i.e., carrier frequency  1  of  FIG. 3 ). In the present example, the first carrier frequency is chosen to be two kiloHertz (2 kHz). 
     The principle of modulation is that the original image will be moved forward by the amount f 1 +f 2 . In addition, the original image is mirrored and will be appear at f 1 −f 2 . The frequency f 1  is the starting frequency of the original spectrum. The frequency f 2  is the frequency that is used to multiply with. 
     Modulator  330  multiplies the original spectrum  410  by a sine wave having the fixed frequency of two kiloHertz (2 kHz).  FIG. 7  illustrates a graph  700  showing the magnitude of the modulated audio spectrum versus frequency. The graph  700  of  FIG. 7  shows the effect of modulating the original spectrum  410  at the modulation frequency of two kilohertz (2 kHz). 
     The modulator  330  then provides the modulated audio spectrum to first filter block  335 . First filter block  335  removes the unwanted frequencies from the modulated audio spectrum. First filter block  335  comprises a low pass filter (not shown in  FIG. 3 ) and a high pass filter (not shown in  FIG. 3 ). 
     The low pass filter of first filter block  335  filters out the high frequency portion of the modulated spectrum. The graph  800  of  FIG. 8  illustrates the filter characteristic  810  of the low pass filter. The filter characteristic  810  of the low pass filter preserves the low frequency portion of the modulated spectrum. The filter characteristic  810  filters out the high frequency portion of the modulated spectrum of the alias. Removing the high frequency portion of the modulated spectrum leaves the low frequency modulated spectrum as shown in  FIG. 9 . 
     Then the high pass filter of first filter block  335  filters out the low frequency portion of the modulated spectrum. The graph  900  of  FIG. 9  illustrates the filter characteristic  910  of the high pass filter. The filter characteristic  910  of the high pass filter preserves the high frequency portion of the modulated spectrum. The filter characteristic  910  filters out the low frequency portion of the modulated spectrum below approximately two thousand Hertz (2.0 kHz). Removing the low frequency portion of the modulated spectrum leaves the frequency modulated spectrum with the shape  1010  shown in  FIG. 10 . 
       FIG. 10  illustrates a graph  1000  that shows the effect of applying both the low pass filter and the high pass filter of first filter block  335  to the modulated spectrum shown in  FIG. 7 . The filtered modulated spectrum  1010  is then sent through gain unit  340 . Gain unit  340  attenuates the spectrum  1010  slightly and then sends the attenuated spectrum  1010  to adder  325 . Adder  325  adds the attenuated version of spectrum  1010  to the original spectrum  410  from delay unit  320 .  FIG. 11  illustrates a graph  1100  that shows the result of adding the attenuated version of spectrum  1010  to the original input spectrum  410 . The combined spectrum ( 410  and  1010 ) now reaches up to approximately five thousand four hundred kiloHertz (5.4 kHz). 
     In the example above the gain unit  340  was set to attenuate the filtered modulated spectrum  1010 . In alternate advantageous embodiments of the invention the gain unit  340  could be used to amplify the filtered modulated spectrum  1010 . That is, the gain unit  340  may be used to increase or decrease the magnitude of the filtered modulated spectrum  1010 . 
     The other two branches of harmonic enhancer  260  operate in the same fashion. In the second branch that comprises modulator  345 , second filter block  350  and gain unit  355 , modulator  345  modulates the original spectrum  410  with a second carrier frequency (i.e., carrier frequency  2  of  FIG. 3 ). The second carrier frequency in this example is larger than the first carrier frequency. The second branch adds an additional filtered modulated spectrum  1210  as shown in  FIG. 12 . 
     In the third branch that comprises modulator  360 , third filter block  365  and gain unit  370 , modulator  360  modulates the original spectrum  410  with a third carrier frequency (i.e., carrier frequency  3  of  FIG. 3 ). The third carrier frequency in this example is larger than the second carrier frequency. The third branch adds an additional filtered modulated spectrum  1220  as shown in  FIG. 12 . 
     The graph  1200  of  FIG. 12  shows the resulting composite of the original spectrum  410 , the first filtered modulated spectrum  1010  from the first branch (the first alias), the second filtered modulated spectrum  1210  from the second branch (the second alias), and the third filtered modulated spectrum from the third branch (the third alias). As previously mentioned, the harmonic enhancer  260  of the present invention may have more than three branches or fewer than three branches. 
       FIG. 13  illustrates a graph of signal magnitude (in decibels) versus frequency (in kiloHertz) for the audio spectrum of the audio signal showing the effect of applying harmonic enhancement to the original audio signal. Line  1310  represents the audio spectrum for the original audio signal  410  after upsampling and low pass filtering.  FIG. 13  shows that line  1310  is substantially flat for the frequencies above three thousand four hundred Hertz (3,400 Hz). 
     Line  1320  represents the audio spectrum for the filtered modulated spectrum  1010  (the first alias). Line  1330  represents the audio spectrum for the filtered modulated spectrum  1210  (the second alias). Line  1340  represents the audio spectrum for the filtered modulated spectrum  1220  (the third alias). Line  1350  represents the final audio spectrum that results when the three harmonic spectra are added to the original audio signal  1310 . 
     It is clear that the three added harmonic spectra have to be significantly weaker than the original audio signal  1310 . To obtain an appropriate final audio spectrum  1350  the frequency response of the speaker  280  must be known and taken into account in setting the gain of the added audio spectra. For example, if an added audio spectrum has to be twenty decibels (20 dB) less than the original audio signal  1310  at a frequency of four kiloHertz (4 kHz), and if the frequency response of the speaker  280  shows an increase of twelve decibels (12 dB) at the same frequency of four kiloHertz (4 kHz), then the gain of the added audio spectrum should be a negative thirty two decibels (−32 dB). 
     As shown in  FIG. 13 , the peak at the front of each of the added harmonic spectra ( 1320 ,  1330 ,  1340 ) is much weaker than the corresponding peak was in the original audio spectrum ( 1310 ). This is done to prevent resonant effects. 
     In each of the filter blocks ( 335 ,  350 ,  365 ) a combination of a high pass filter and a low pass filter was used instead of a band pass filter. This structure was selected so that the cutoff frequencies overlap. Overlapping the cutoff frequencies minimizes the number of taps required in the filters. By shifting the high pass filter of the filter block more to the left, the peak at the front of the spectrum may be reduced in order to limit the resonant effect. 
     It is understood, however, that the harmonic enhancer of the invention is not limited to using a separate high pass filter and a separate low pass filter in the filter blocks. In an alternate embodiment of the invention the high pass filter and the low pass filter of each filter block may be replaced with a band pass filter. 
       FIG. 14  illustrates a flow chart  1400  showing the steps of a first portion of an advantageous embodiment of the method of the present invention. In the first step an audio signal is received in the baseband integrated circuit  230  of a receiver handset  200  (step  1410 ). If the audio signal is an analog audio signal then the audio signal is converted to a digital audio signal (step  1420 ). The digital audio signal is then processed in a generic digital signal processor  250  that comprises a harmonic enhancer  260  of the present invention (step  1430 ). 
     In the harmonic enhancer  260  the original audio spectrum  410  is upsampled in a times two (2) upsample unit  310  (step  1440 ). This doubles the signal bandwidth (e.g., from four kiloHertz (4 kHz) to eight kiloHertz (8 kHz)) and creates an alias spectrum  510  of the original spectrum. The alias spectrum  510  is located in the higher frequencies of the spectrum (e.g., between four kiloHertz (4 kHz) and eight kiloHertz (8 kHz). The resulting spectrum is then sent through a low pass filter  315  to remove the alias spectrum  510  (step  1450 ). The resulting spectrum (e.g., the original audio spectrum  410  now in an eight kiloHertz (8 kHz) bandwidth) is modulated in modulator  330  with a first carrier frequency (e.g., two kiloHertz (2 kHz)) (step  1460 ). The method then proceeds to step  1510  of  FIG. 15 . 
       FIG. 15  illustrates a flow chart  1500  showing the steps of a second portion of an advantageous embodiment of the method of the present invention. The method proceeds from step  1460  of  FIG. 14 . The modulated spectrum from modulator  330  is then sent to first filter block  335  and filtered in a low pass filter (e.g., filter characteristic  810 ) to remove unwanted high frequency portions of the spectrum (step  1510 ). The resulting spectrum is then filtered in a high pass filter (e.g., filter characteristic  910  of first filter block  335  to remove unwanted low frequency portions of the spectrum (step  1520 ). 
     Then the resulting filtered modulated spectrum  1010  is sent to gain unit  340  and the amplitude of spectrum  1010  is adjusted (step  1530 ). Usually the amplitude of spectrum  1010  is reduced to create an attenuated version of spectrum  1010 . After amplitude adjustment, the filtered modulated spectrum  1010  is sent to adder unit  325 . Spectrum  1010  is then added to a delayed version of the original spectrum  410  that has been sent through delay unit  320  (step  1540 ). 
     At the same time that spectrum  1010  is created, a second spectrum  1210  is similarly created and using modulator  345 , a second carrier frequency, second filter block  350  and gain unit  355 . At the same time that spectrum  1010  and spectrum  1210  are created, a third spectrum  1220  is similarly created and using modulator  360 , a third carrier frequency, third filter block  365  and gain unit  370 . The creation of additional spectra and the addition of the additional spectra in adder unit  325  are designated with reference numeral  1550 . The harmonically enhanced audio signal of the present invention is then output to be sent to speaker  280  (step  1560 ). 
     The addition of one harmonic spectrum (e.g., spectrum  1010 ) to the original audio spectrum ( 410 ) produces a sharper sound. The sound is further improved by repeating the modulation on different frequencies (i.e., adding spectrums  1210  and  1220 ). Modulated sounds have a rather sharp sound. The combined spectrum may be made closer to the original sounds by attenuating the modulated sounds. The addition of the modulated sounds to the original audio provides a subjective improvement to the audio quality. That is, not everyone may agree that the additions are enhancements to the original audio quality. For this reason, the receiving system  200  of the present invention is provided with a switch (not shown) that will selectively enable and disable the harmonic enhancer  260  as directed by the end user. 
     The generic digital signal processor (GenDSP)  250  in baseband integrated circuit  230  can perform the function of harmonic enhancer  260  in real time. For example, if baseband integrated circuit  230  is implemented by the SC14428 baseband chip, the SC14428 baseband chip is able to sample audio signals with a sixteen kiloHertz (16 kHz) frequency. With a sample frequency of sixteen kiloHertz (16 kHz), the ADPCM decoder  240  of the SC14428 baseband chip can process the eight kiloHertz (8 kHz) sampled data of the radio frequency (RF) interface by means of a software buffer (not shown). The SC14428 baseband chip can process a maximum effective audio band of one hundred Hertz (100 Hz) to six thousand eight hundred Hertz (6,800 Hz). 
     For best results, the parameters of the harmonic enhancer  260  should be adjusted to match the hardware (e.g., speaker circuitry) that produces the sound. For example, a speaker can reproduce certain frequencies harder or softer. This may cause the effect of the harmonic enhancer  260  to have less than the desired effect. This phenomenon occurs due to the acoustic properties of the speaker and the speaker-cabinet. This problem may be minimized by adjusting the parameters of the harmonic enhancer  260  using acoustic measurements taken of the speaker  280  in its final housing. The measurements of the speaker  280  are best performed in an acoustically “dead” room where there is no noise interference. 
     The harmonic enhancer  260  of the present invention may be used in any audio application in which digital audio signals are transmitted over a limited bandwidth. In one advantageous embodiment of the invention the received signal does not have to be a digital signal. The received signal may be an analog signal that is digitized before it is played for the receiving listener (i.e., a digital enhancement of the original analog signal). 
     The harmonic enhancer  260  of the present invention may be used in digital cordless telephone handsets. The digital cordless telephone handsets may be compliant with the Desktop Computer Telephone Integration (DCTI) standard, the Digital Enhanced Cordless Telecommunication (DECT) standard, the Personal Handyphone System (PHS) standard, and many other similar types of standards. 
     The harmonic enhancer  260  of the present invention may be used in digital cellphone telephone handsets. The digital cellphone telephone handsets may be compliant with the Global System for Mobile Communications (GSM) standard, the Code Division Multiple Access (CDMA) standard, the Universal Mobile Telecommunications System (UMTS) standard, and many other similar types of standards. 
     The harmonic enhancer  260  of the present invention may be used in digital satellite telephone handsets and other similar communications systems. In addition, the harmonic enhancer  260  of the present invention may be used in digital corded telephone handsets, digital speaker phones, digital intercom systems, and walkie talkie systems. The harmonic enhancer  260  of the present invention may also be used in digital “voice over Internet Protocol” (VoIP) systems and Internet telephony. 
     The harmonic enhancer  260  of the present invention may be used in any type of device that utilizes digitally stored voice playback, such as digital audio telephone answering machines, voicemail, automated audio response systems, digital memorandum recorders, and digital audio talking toys. 
     Although the present invention has been described with an exemplary embodiment, various changes and modifications may be suggested to one skilled in the art. It is intended that the present invention encompass such changes and modifications as fall within the scope of the appended claims.