Apparatus for generating harmonics in an audio signal

Information

  • Patent Grant
  • 6792115
  • Patent Number
    6,792,115
  • Date Filed
    Friday, November 17, 2000
    24 years ago
  • Date Issued
    Tuesday, September 14, 2004
    20 years ago
Abstract
An apparatus for generating harmonics in an audio signal with an adder stage, which can add signals from a first signal line and a second signal line, such that an audio signal with harmonics can be tested at the output of the adder stage and such that a filter device and a nonlinear circuit device are situated in the second signal line, characterized in that a corner frequency of the filter device is adjustable.
Description




BACKGROUND OF THE INVENTION




The invention relates to an apparatus and method for generating harmonics in an audio signal.




Methods and circuits for generating harmonics are used in devices for acoustic reproduction, such as television receivers, radio receivers, and stereo systems, to compensate the frequency response of the loudspeakers, in order to improve acoustic reproduction and also to prevent the device or the system from being overdriven.




A critical element in a device for acoustic reproduction is the loudspeaker, whose acoustic pressure, below a structure-based limit frequency, drops about 40 db per decade. This corresponds to the transmission function of a second-order filter. On the other hand, bass reflex and transmission line loudspeakers have transmission functions corresponding to a filter of higher order. The lower limit frequency typically lies between about 50 Hz and 200 Hz. The lower the limit frequency of a loudspeaker, the more expensive it is to produce. Consequently, economical devices such as TV sets and portable radio receivers are equipped with simpler loudspeakers, whose lower limit frequency is relatively high. To improve acoustic reproduction in the lower frequency range, the limit frequency is displaced downward by pre-amplifying the low frequencies. However, this can cause the final amplifier and the loudspeakers to be overdriven. To prevent the final amplifiers or loudspeakers from being overdriven and possibly even being destroyed, the output signal of the bass amplifier is fed back in such a way that the amplification of the lower frequencies is reduced for a large output signal. For example, U.S. Pat. No. 5,305,388 entitled “Bass Compensation Circuits For Use In Sound Reproduction Device” discloses such a method.




U.S. Pat. No. 5,359,665 entitled “Audio Bass Frequency Enhancement” discloses a circuit arrangement in which the audio signal is conducted via a first path directly to the first input of an adder, and conducted via a second path through a low-pass filter and an amplifier with variable amplification to a second input of the adder. The output of the amplifier is fed back via a signal level detector to its control input. This measure reduces overdrive of the final amplifier.




From psychoacoustics, it is known that a person can still clearly determine the fundamental frequency of a tone even if the fundamental frequency is not present in the spectrum, but only harmonics of the fundamental frequency. This psychoacoustic effect is utilized in that harmonics of the fundamental frequency are generated and are conducted to a loudspeaker whose limit frequency lies above this fundamental frequency. A listener thus thinks that he is hearing this low fundamental frequency, even though the loudspeaker does not radiate it. For example, a listener thinks that he is hearing a tone of 50 Hz when the loudspeaker in fact does not transmit this low tone, but only a tone of 250 Hz and a tone of 300 Hz. Advantageously, a listener subjectively perceives the difference of 50 Hz.




This effect is utilized when qualitatively simple loudspeakers with a high lower limit frequency of for example 120 Hz, are supposed to transmit for example a signal of 60 Hz. One then generates harmonics of the 60 Hz signal whose difference amounts to 60 Hz. The listener then actually thinks he is hearing a 60 Hz tone, although this tone is not radiated by the loudspeaker.




To generate the harmonics electronic circuits are required to determine the fundamental frequency in a mixture of audio signals, and to extract it in order to generate the harmonics of this fundamental frequency.




U.S. Pat. Nos. 5,668,885 and 5,771,296 describe the generation of harmonics by forming absolute values by a rectifier arrangement.




U.S. Pat. Nos. 4,150,253 and 4,700,390 describe the generation of harmonics by clipping (i.e., cutting off the amplitude of the fundamental frequency of the audio signal above a certain level).




In all these documents, filters with fixed corner frequencies are used to select the signals whose harmonics are to be generated.




If there is more than one signal in the selected frequency range (as is the case for real audio signals), which are composed of a frequency spectrum, then not only the harmonics of the existing signals are generated but also harmonics with an undesirable frequency, which consist of the sum of all the existing signal frequencies and their multiples. The result of this is that the tone finally radiated by the loudspeaker sounds very muddy.




Therefore, there is a need for an improved technique for generating harmonics.




SUMMARY OF THE INVENTION




Briefly, according to an aspect of the present invention, a system for generating harmonics in an audio signal includes a filtering device that is responsive to an audio input signal, and determines a dominant fundamental frequency component within the audio input signal and provides a filtered audio signal indicative thereof. A non-linear unit (e.g., a multiplier) responsive to the filtered audio signal generates harmonic frequency components of the dominant fundamental frequency and provides a non-linear unit output signal indicative thereof to a first bandpass filter that provides a first bandpass filtered signal. A summer sums the audio input signal and the first bandpass filtered signal to provide a system output signal that includes harmonics of the dominant frequency component.




In one embodiment, a system delimits as precisely as possible the signal frequency whose harmonics are to be generated with a variable filter (e.g., a band-pass filter).




A second embodiment generates harmonics by using the nth power (e.g., two) of the input signal to generate the (n−1) harmonic. If n is equal to two, the fundamental frequency of the input signal is squared and the first harmonic is generated. Fundamental frequency is here understood to designate the dominating frequency contained in the audio signal, within a frequency range of for example less than 120 Hz. Generating harmonics by potentiation is much “cleaner” compared to the known clipping or rectifying process. The amplitude of the signal is corrected before or after potentiation.




Advantageously, a system of the present invention creates an improved audio impression for a listener when he hears an audio signal that is radiated by loudspeakers with a relatively high lower corner frequency.




These and other objects, features and advantages of the present invention will become apparent in light of the following detailed description of preferred embodiments thereof, as illustrated in the accompanying drawings











BRIEF DESCRIPTION OF THE DRAWING





FIG. 1

is a block diagram illustration of a first inventive circuit arrangement with a device for delimiting the lower fundamental frequency;





FIG. 2

is a somewhat more detailed block diagram illustration of a circuit embodiment than

FIG. 1

for generating harmonics in an audio signal;





FIG. 3

is a block diagram illustration of a second inventive circuit arrangement with a device for potentiating the fundamental frequency signal;





FIG. 4

is a block diagram illustration of

FIG. 3

in more detail; and





FIG. 5

is a block diagram illustration of variant of the arrangement of FIG.


4


.











DETAILED DESCRIPTION OF THE INVENTION





FIG. 1

illustrates a circuit arrangement


10


, in which an audio input signal si on a line


20


is conducted to the input terminal of an adder stage


22


. The output of the adder stage


22


is connected for example to a loudspeaker or to an amplifier (not shown). The input signal si is also conducted via a filter device, here a low-pass filter


24


or a band-pass filter, to a nonlinear circuit unit


26


, which generates harmonics from the filtered signal. The filter arrangement is used to determine the dominating fundamental frequency in the input signal si on the line


20


. From this, the nonlinear circuit


26


generates harmonics and conducts these to a second input terminal of the adder stage


22


. It is essential that the corner frequency fc of the filter


24


is adjustable.





FIG. 2

illustrates a somewhat more detailed circuit embodiment than

FIG. 1

for generating harmonics in an audio signal. The filter includes a series circuit of a first low-pass filter


32


, a following second low-pass filter


34


and a subsequent high-pass filter


36


. The signals at the respective inputs are designated by s


1


, s


2


, and s


3


. The high-pass filter


36


provides an output signal s


4


that is input to a nonlinear circuit unit


38


, which provides an output signal s


5


to a bandpass filter


40


. The filter


40


generates a bandpassed output signal s


6


, which may be amplified by a gain factor g and input to an adder stage


42


.




The low-pass filter


32


has a fixed corner frequency (e.g., 200 Hz). The low-pass filter


34


has a variable corner frequency fc. The high-pass filter


36


has for example a constant corner frequency of 50 Hz, or of k*fc, with k chosen to be less than one. The band-pass filter


40


has for example a center frequency that is proportional to fc.




The corner frequency fc of the low-pass filter


34


can be adjusted in response to the signals s


2


and s


3


. Specifically, the signals s


2


and s


3


are input to a comparator


44


that provides a control signal on a line


46


that sets the corner frequency fc for the low pass filter


34


.




The circuit arrangement of

FIG. 2

functions as follows. The low frequencies within the spectrum of the input signal s


1


are pre-selected in the low-pass filter


32


. Further filtering takes place in the low-pass filter


34


. The corner frequency fc is adjusted so that the following holds:








s




3


=


q*s




2


, where 0


<q<


1.






(Remark: “*” designates multiplication)




In the above formula, s


3


and s


2


can designate the signal amplitudes of the signals s


3


and s


2


or also their signal energies.




The corner frequency fc is thus adjusted so that a certain portion q of the signal s


2


is present at the output of the low-pass filter


34


. Only the lowest frequency components of the input signal si (note si=s


1


) can thus pass. Other interfering signal frequencies are filtered out.




The corner frequency fc can be determined for example by the following:







fc=fc+df,


if


s




3


<


q*s




2










fc=fc−df,


if


s




3


>


q*s




2


.






Here df determines the rate of conversions of the rule and is advantageously chosen equal to or less than 1 Hz. That is, the low frequency range of the audio input signal si is queried (e.g., in 1 Hz steps), and the frequency is determined at which the signal has the greatest amplitude or energy. This frequency is then the desired fundamental frequency, from which harmonics will be generated in the nonlinear unit


38


.




The signal s


3


is high-pass filtered in the high-pass filter


36


. The corner frequency of the high-pass filter


36


can either be constant or selected as a function of fc. Undesired frequencies in the signal s


5


are removed by the band-pass filter


40


. The center frequency of the band-pass filter


40


may be adjusted as a function of the corner frequency fc. For example, if the nonlinear unit


38


generates mainly signals with the first harmonic (i.e., double the signal frequency of the fundamental) the center frequency can be chosen such that fbp=2*fc, where fbp is the center frequency of the bandpass filter


40


.




The signal s


6


at the output of the band-pass filter BP


2


is preferably amplified by a factor g before it is added to the signal si in the adder stage


42


.





FIG. 3

illustrates a circuit arrangement


50


in which a band-pass filter arrangement


52


that includes a low-pass filter


54


and a subsequent high-pass filter


56


, is situated in the signal branch which does not directly lead to adder stage


58


. The band-pass filter


52


provides an output signal s


4


that represents the fundamental frequency. The filter arrangement


52


preferably is the one shown in connection with

FIGS. 1 and 2

.




The signal s


4


is squared in the nonlinear unit


60


to generate the first harmonic. The nonlinear unit


60


provides an output signal s


5


that is input to a band-pass filter


62


, which provides a bandpassed signal s


6


. The bandpassed signal s


6


is conducted to the adder stage


58


, either directly or first multiplied by a factor g. Rather than being squared, the signal s


4


may be raised by a power of 3 or 4, or a higher integer value.





FIGS. 4 and 5

are alternative embodiments of the circuit arrangement of

FIG. 3

, because there the signal is normalized before (

FIG. 4

) or after (

FIG. 5

) potentiation.




Referring to

FIG. 4

, in this embodiment the signal s


2


is input to an RMS detector


72


. The output from the RMS detector


72


is input to a divider stage


74


, which also receives the output signal s


3


from unit


76


. The divider stage


74


divides its output signal s


3


by the value of the output signal from the RMS detector


72


. The divider stage


74


is followed by a limiter


78


, whose output is connected to a band-pass filter


80


.




By being normalized with the RMS value of the signal which is calculated by the RMS detector


72


, the signal s


3


again recovers its original amplitude, in accordance with








s




4


=(


s




2




2


)/


RMS


(


s




2


).






The detector RMS has a time constant tau of for example 0.2 seconds. Since the amplitude of s


3


under some circumstances can rise much faster than the RMS value, which has a higher time constant, very high values can occur in the signal s


4


. Consequently, the values of the signal s


4


are limited to a permissible value in the limiter


78


. The generated harmonics are again limited in the band-pass filter


80


with the center frequency fbp


2


, and then in the adder stage AD are added to the audio signal si.




In the embodiments set forth in

FIGS. 3-5

, the nonlinear unit


76


preferably squares the signal s


2


to generate first-order harmonics. Therefore, in these embodiments the center frequency fbp


2


of the second band-pass filter (


62


,


80


) should be chosen to be twice the center frequency fbp


1


of the first band-pass filter (i.e., BPL


1


).




However, higher-order harmonics can be generated just as well if the nonlinear unit is appropriately designed for this. For potentiating by a factor of 3, one would have to make fbp


2


=fbp


1


*3, since the frequency was tripled.




The framework of the invention also comprises parallel connection of several of the circuits presented above, in order to simultaneously generate several harmonics (e.g., the first and the second harmonic).




Although the present invention has been shown and described with respect to several preferred embodiments thereof, various changes, omissions and additions to the form and detail thereof, may be made therein, without departing from the spirit and scope of the invention.



Claims
  • 1. A system for generating harmonics in an audio signal, comprising:a filtering device that is responsive to an audio input signal, and determines a dominant fundamental frequency component within said audio input signal and provides a filtered audio signal indicative thereof; a non-linear unit responsive to said filtered audio signal, that generates harmonic frequency components of the dominant fundamental frequency and provides a non-linear unit output signal indicative thereof; a first bandpass filter that is responsive to said non-linear unit output signal and provides a first bandpass filtered signal; and a summer that sums said audio input signal and said first bandpass filtered signal to provide a system output signal that includes harmonics of the dominant frequency component.
  • 2. The system of claim 1, wherein said non-linear unit comprises means for squaring said filtered audio signal and for providing said non-linear output signal.
  • 3. The system of claim 1, wherein said non-linear unit comprises means responsive to said filtered audio output signal for raising the power of said filtered audio signal by an integer value.
  • 4. The system of claim 1, wherein said filtering device comprises:a first low pass filter that is responsive to said audio input signal and provides a first filtered signal; an adjustable second low pass filter that is responsive to said first filtered signal and a command signal, wherein said command signal sets the break frequency for said adjustable second low pass filter that provides a second filtered signal; and a high pass filter that is responsive to said second filtered signal and provides a signal indicative of said filtered audio signal.
  • 5. The system of claim 4, wherein said adjustable second low pass filter comprises means responsive to said first filtered signal and said second filtered signal, for determining said command signal.
  • 6. The system of claim 5, wherein said means for determining said command signal comprises a comparator that compares said first and second filtered signals.
  • 7. The system of claim 5, wherein said first bandpass filter is responsive to said command signal to set the center frequency of the pass band of said first bandpass filter.
  • 8. The system of claim 1, wherein said filtering device comprises:a second band pass filter that is responsive to said audio input signal and provides a second bandpass filtered signal; an RMS detector that is responsive to said second bandpass filtered signal and provides an RMS output signal; a multiplier that is responsive to said second bandpass filtered signal, and raises the power of said second bandpass filtered signal and provides a multiplier output signal indicative thereof; and a divider that divides said multiplier output signal by said RMS output signal and provides a divider output signal that is indicative of said filtered audio signal.
  • 9. A system for generating harmonics in an audio signal, comprising:a filtering device that is responsive to an audio input signal, and determines a dominant fundamental frequency component within said audio input signal and provides a filtered audio signal indicative thereof; a multiplier circuit that is responsive to said filtered audio signal and raises the power of said filtered audio signal by an integer value of at least two to generate harmonic frequency components of the dominant fundamental frequency and provide a multiplier circuit output signal indicative thereof; means for filtering responsive to said multiplier circuit output signal for providing a first bandpass filtered signal; and a circuit that sums said audio input signal and said first bandpass filtered signal to provide a system output signal that includes harmonics of the dominant frequency component.
  • 10. A system for generating harmonics in an audio signal, comprising:a filtering device that is responsive to an audio input signal, and determines a dominant fundamental frequency component within said audio input signal and provides a filtered audio signal indicative thereof; a multiplier responsive to said filtered audio signal, that generates harmonic frequency components of the dominant fundamental frequency and provides a multiplier output signal indicative thereof; a first bandpass filter responsive to said multiplier output signal for providing a first bandpass filtered signal; and a summing circuit that sums said audio input signal and said first bandpass filtered signal to provide a system output signal that includes harmonics of the dominant frequency component.
  • 11. The system of claim 10, whereinsaid multiplier raises the power of said filtered audio signal by an integer value; and said filtering device comprises (i) a first low pass filter that is responsive to said audio input signal and provides a first filtered signal; (ii) an adjustable second low pass filter that is responsive to said first filtered signal and a command signal, wherein said command signal sets the break frequency for said adjustable second low pass filter that provides a second filtered signal; and (iii) a high pass filter that is responsive to said second filtered signal and provides a signal indicative of said filtered audio signal.
  • 12. The system of claim 11, wherein said adjustable second low pass filter comprises means responsive to said first filtered signal and said second filtered signal, for determining said command signal.
  • 13. The system of claim 12, wherein said means for determining said command signal comprises a comparator that compares said first and second filtered signals.
  • 14. The system of claim 10, wherein said first bandpass filter is responsive to said command signal to set the center frequency of the pass band of said first bandpass filter.
Priority Claims (1)
Number Date Country Kind
199 55 696 Nov 1999 DE
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Number Name Date Kind
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5668885 Oda Sep 1997 A
5754666 Nakagawa May 1998 A
5771296 Unemura Jun 1998 A
5930373 Shashoua et al. Jul 1999 A
6111960 Aarts et al. Aug 2000 A
6134330 De Poortere et al. Oct 2000 A
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