Digital equalizer apparatus enabling separate phase and amplitude characteristic modification

Claims

What is claimed is:

1. A digital equalizer apparatus comprising:

amplitude/frequency input means for providing amplitude/frequency characteristic data representing an arbitrary amplitude/frequency characteristic;

phase data input means for providing data for establishing a phase/frequency characteristic, said data being in a category selected from a group of categories of data which includes phase/frequency characteristic data, group delay characteristic data, amplitude/frequency characteristic data, and data expressing a resonance condition of a predetermined type of electrical circuit;

phase/frequency operational means for computing phase/frequency characteristic data based upon said data from said phase data input means;

transfer function operational means for operating on said amplitude/frequency characteristic and phase/frequency characteristic data to derive transfer function data representing a transfer function;

inverse Fourier transform means for operating on said transfer function data to derive impulse response characteristic data representing an impulse response characteristic determined by said transfer function;

finite impulse response filter means;

signal input means for transferring to said finite impulse response filter means an input audio signal as a train of digital samples;

signal output means for receiving said audio signal after frequency characteristic modification of said audio signal by said finite impulse response filter means, and for transferring the modified audio signal to an external system; and,

setting means operable for establishing a set of filter coefficients for said finite impulse response filter means having respective values determined by said impulse response characteristic.

2. A digital equalizer apparatus according to claim 1, in which said data from said phase data input means represent a group delay characteristic, and in which said phase/frequency operational means comprises integrator means for integrating said group delay characteristic data with respect to frequency, for deriving said phase/frequency characteristic to be supplied to said transfer function operational means.

3. A digital equalizer apparatus according to claim 1, in which said data from said phase data input means represent a group delay characteristic, and in which said phase/frequency operational means comprises means for deriving an average value of group delay over a predetermined frequency range and for redefining said group delay characteristic as an amended group delay characteristic representing deviations from said average value, and integrator means for integrating said amended group delay characteristic data with respect to frequency, thereby deriving said phase/frequency characteristic to be supplied to said transfer function operational means.

4. A digital equalizer apparatus according to claim 1, in which said data from said phase data input means represent an amplitude/frequency characteristic, and in which said phase/frequency operational means comprises means for computing the Hilbert transform of said amplitude/frequency characteristic from the phase data input means to thereby derive said phase/frequency characteristic to be supplied to said transfer function operational means.

5. A digital equalizer apparatus according to claim 1, in which said finite impulse response filter means comprises a plurality of finite impulse response filters, and further comprising:

a plurality of digital band-pass filters for dividing said digital sample signal from said signal input means into a plurality of frequency bands; and

a plurality of down-sampling sections for receiving respective band-divided output signals from said band-pass filters for reducing the sampling frequency of said band-divided output signals by respectively differing reduction factors, and supplying resultant digital sample signals to respective ones of said plurality of finite impulse response filters.

6. A digital equalizer apparatus according to claim 5, and further wherein said transfer function operational means derives, based on said amplitude/frequency characteristic and phase/frequency characteristic data supplied thereto, respectively different transfer functions for each of said plurality of finite impulse response filters, and wherein each of said transfer functions is defined within a frequency range having an upper frequency limit which is lower than the sampling frequency of the digital sample signal supplied from the corresponding one of said down-sampling sections to the corresponding one of said finite impulse response filters, with impulse response characteristics respectively corresponding to said plurality of transfer functions being derived by said inverse Fourier transform means and with sets of filter coefficients in accordance with respective ones of said impulse response characteristics being established for respectively corresponding ones of said finite impulse response filters by said setting means.

7. A digital equalizer apparatus according to claim 6, in which said signal output means comprises a plurality of output means respectively coupled to receive output signals from corresponding ones of said finite impulse response filters.

8. A digital equalizer apparatus according to claim 5, and further comprising:

a plurality of up-sampling sections coupled to receive output signals for respective ones of said finite impulse response filters, for restoring the sampling frequencies of said output signals to that of said digital sample signal from said signal input means; and,

addition means for combining respective output signals produced from said up-sampling sections, for producing an output signal to be supplied to said signal output means.

9. A digital equalizer apparatus according to claim 5, and further comprising phase compensation means for executing phase compensation of said phase/frequency characteristic data from said phase/frequency operational means such that phase values of said phase/frequency characteristic at respective ones of guard band frequencies, each separating an upper limit frequency of one of said transfer functions from a higher frequency within the frequency range of another one of said transfer functions, are made mutually identical.

10. A digital equalizer apparatus according to claim 1, and further comprising:

amplitude and phase characteristic analysis means for analyzing the frequency characteristics of a measurement signal produced from a microphone in response to an acoustic signal generated within a sound field, to derive an amplitude deviation/frequency characteristic and a phase/frequency deviation characteristic of said measurement signal;

amplitude/frequency characteristic compensation means for operating on said amplitude/frequency characteristic data from said amplitude/frequency characteristic input means and said amplitude deviation/frequency characteristic in combination, to derive compensated amplitude/frequency characteristic data to be supplied to said transfer function operational means for use in computing said transfer function; and,

phase/frequency characteristic compensation means for operating on said phase/frequency characteristic data from said phase/frequency operational means and said phase/frequency deviation characteristic in combination, to derive compensated phase/frequency characteristic data to be supplied to said transfer function operational means for use in computing said transfer function;

said digital equalizer apparatus being selectively operable in a measurement condition in which said measurement signal is supplied to said signal input means to be transferred to said amplitude and phase characteristic analysis section, a transfer function is computed from said compensated amplitude/frequency characteristic and phase/frequency characteristic data, and filter coefficients corresponding to said transfer function are established for said finite impulse response filter means by said setting means, and a normal condition in which an audio signal is supplied to said signal input means to be transferred by said signal input means to said finite impulse response filter means.

11. A digital equalizer apparatus according to claim 10, in which said acoustic signal is an impulse signal.

12. A digital equalizer apparatus according to claim 1, further comprising:

test signal generating means for generating a test signal to be transferred by said signal output means to an external audio system including a loudspeaker, for producing a corresponding measurement signal by a microphone disposed within a sound field together with said loudspeaker;

amplitude and phase characteristic analysis means for analyzing the frequency characteristics of said measurement signal, to derive an amplitude deviation/frequency characteristic and a phase/frequency deviation characteristic of said measurement signal;

amplitude/frequency characteristic compensation means for operating on said amplitude/frequency characteristic data from said amplitude/frequency characteristic input means and said amplitude deviation/frequency characteristic in combination, to derive compensated amplitude/frequency characteristic data to be supplied to said transfer function operational means for use in computing said transfer function; and,

phase/frequency characteristic compensation means for operating on said phase/frequency characteristic data from said phase/frequency operational means and said phase/frequency deviation characteristic in combination, to derive compensated phase/frequency characteristic data to be supplied to said transfer function operational means for use in computing said transfer function;

said digital equalizer apparatus being selectively operable in a measurement condition in which said test signal is supplied to said signal output means while said measurement signal is supplied to said signal input means to be transferred to said amplitude and phase characteristic analysis section, and in which a transfer function is computed from said compensated amplitude/frequency characteristic and phase/frequency characteristic data, and filter coefficients corresponding to said transfer function are established for said finite impulse response filter means by said setting means, and a normal condition in which an audio signal is supplied to said signal input means for transfer to said finite impulse response filter means.

13. A digital equalizer apparatus according to claim 12, and further comprising memory means for sequentially storing successive ones of said digital samples transferred from said signal input means while said measurement signal is being produced, and for supplying data thus stored to said amplitude and phase characteristic analysis means for deriving said amplitude deviation/frequency characteristic and phase/frequency deviation characteristic.

14. A digital equalizer apparatus according to claim 13, and further comprising decision means for detecting when an amplitude of said measurement signal exceeds a predetermined value and functioning, when said detection occurs, to control said memory means to initiate said sequential storage of digital samples.

15. A digital equalizer apparatus according to claim 14, and further comprising synchronized addition means controlled by said decision means, each time that said measurement signal amplitude is detected as having exceeded said predetermined value thereof, for executing a process of sequential read-out of stored digital samples from said memory means in synchronism with outputs of successive digital samples form said signal input means, addition of said read-out samples to corresponding digital samples transferred from said signal input means, and storage of results obtained from said addition in said memory means.

16. A digital equalizer apparatus according to claim 15 in which output signals produced from said signal output means of the digital equalizer apparatus are applied to an audio system which includes a loudspeaker located within said sound field, and in which said digital equalizer apparatus further comprises:

test signal generating means functioning during said measurement condition to sequentially generate test signals during successive time intervals, said test signals being transferred by said signal output means to said audio system for producing corresponding successive acoustic signals from said loudspeaker; and,

time measurement and control means operable when said measurement condition is entered for measuring a time interval which elapses between initiation of a first one of said test signals and a time at which said predetermined value is subsequently detected by said decision means, and for thereafter initiating operation of said synchronized addition means after said measured time interval has elapsed, following each initiation of subsequent ones of said test signals.

17. A digital equalizer apparatus according to claim 15 in which an output signal produced from said signal output means of the digital equalizer apparatus are applied to an audio system which includes a loudspeaker located within said sound field, and in which said digital equalizer apparatus further comprises:

test signal generating means functioning during said measurement condition to sequentially generate test signals during successive time intervals, said test signals being transferred by said signal output means to said audio system for producing corresponding successive acoustic signals from said loudspeaker; and,

time measurement and control means operable when said measurement condition is entered for measuring a time interval which elapses between initiation of a first one of said test signals and a time at which said predetermined value is subsequently detected by said decision means, and for thereafter initiating operation of said synchronized addition means after a fixed time interval which is shorter than said measured time interval has elapsed, following each initiation of subsequent ones of said test signals.

18. A digital equalizer apparatus according to claim 13, further comprising window function means for multiplying measurement signal data that has been stored in said memory means by a predetermined window function, and for storing the results obtained in said memory means, and in which said results are then supplied to said amplitude and phase characteristic analysis means for deriving said amplitude deviation/frequency characteristic and phase/frequency deviation characteristic.

19. A digital equalizer apparatus according to claim 18, in which said window means functions to multiply said stored measurement signal data by a plurality of mutually different window functions which correspond to respectively different time intervals, and in which the respective results of multiplications by these window functions are utilized by said amplitude and phase characteristic analysis means to derive respective amplitude deviation/frequency characteristics which cover mutually different frequency bands, and respective phase/frequency deviation characteristics which cover said mutually different frequency bands.

20. A digital equalizer apparatus according to claim 13, and further comprising sampling frequency reduction means for eliminating a proportion of said digital samples of said measurement signal from said signal input means and for supplying the remaining digital samples to be stored in said memory means.

21. A digital equalizer apparatus according to claim 19, and further comprising phase equalizing means for adding or subtracting fixed values of phase to or from said respective results obtained by multiplication by said respectively different window functions, said fixed values being selected such as to equalize mutually adjacent portions, respectively obtained utilizing different ones of said window functions, of said amplitude deviation/frequency characteristic derived by said amplitude and phase characteristic analysis means, and to equalize mutually adjacent portions, respectively obtained utilizing different ones of said window functions, of said phase deviation/frequency characteristic derived by said amplitude and phase characteristic analysis means.

22. A digital equalizer apparatus according to claim 1, and further comprising:

memory means for storing phase data representing a predetermined phase/frequency characteristic or group delay characteristic together with amplitude data representing a predetermined amplitude deviation/frequency characteristic;

amplitude/frequency characteristic compensation means for operating on said amplitude/frequency characteristic data from said amplitude/frequency characteristic input means and said amplitude data from said memory means, in combination, to derive compensated amplitude/frequency characteristic data to be supplied to said transfer function operational means for use in computing said transfer function; and,

phase/frequency characteristic compensation means for operating on said phase/frequency characteristic data from said phase/frequency operational means and said phase data from said memory means, in combination, to derive compensated phase/frequency characteristic data to be supplied to said transfer function operational means for use in computing said transfer function.

23. A digital equalizer apparatus according to claim 1, and further comprising means for multiplying said impulse response characteristic derived by said inverse Fourier transform means by a predetermined window function, said filter coefficients being respectively established in accordance with results obtained from said window function multiplication operation.

24. A digital equalizer apparatus according to claim 1, in which an output signal produced from said signal output means is applied to an audio system to drive a loudspeaker, and in which said apparatus is operable in a measurement condition in which a measurement signal representing a phase deviation/frequency characteristic and an amplitude deviation/frequency characteristic of said loudspeaker is supplied to said signal input means and a normal condition in which an audio signal is supplied to said signal input means, said apparatus further comprising:

test signal generating means for generating a predetermined form of test signal to be transferred by said signal output means to said audio system for driving said loudspeaker and thereby producing said measurement signal;

loudspeaker phase characteristic analysis means for analyzing said measurement signal for deriving a phase deviation/frequency characteristic and amplitude deviation/frequency characteristic of said loudspeaker;

amplitude/frequency characteristic compensation means for operating on said amplitude/frequency characteristic data from said amplitude/frequency characteristic input means and said amplitude deviation/frequency characteristic in combination, to derive compensated amplitude/frequency characteristic data to be supplied to said transfer function operational means for use in computing said transfer function; and,

phase/frequency characteristic compensation means for operating on said phase/frequency characteristic data from said phase/frequency operational means and said phase deviation/frequency characteristic in combination, to derive compensated phase/frequency characteristic data to be supplied to said transfer function operational means for use in computing said transfer function;

and whereby after derivation of said transfer function in said measurement condition, said filter coefficients are established in accordance with said impulse response characteristic, and said digital equalization apparatus is then set in said normal condition whereby an audio signal is transferred by said signal input means to said finite impulse response filter means.

25. A digital equalizer apparatus comprising:

input means for fixedly supplying data representing an amplitude/frequency characteristic

transform operational means for computing from said amplitude/frequency characteristic data first and second transfer functions respectively corresponding to a low-frequency band and a high-frequency band of a frequency band-separated input audio signal, said transform operational means being operable for selectively computing said first and second transfer functions by a linear phase transform computation method or by a Hilbert transform computation method;

inverse Fourier transform means for deriving first and second impulse response characteristics from said first and second transfer functions respectively;

a first finite impulse response filter coupled to receive said low-frequency band of said frequency band-separated input signal as a train of digital samples;

a second finite impulse response filter coupled to receive said high-frequency band of said frequency band-separated input signal as a train of digital samples;

setting means for establishing first and second sets of filter coefficients for said first and second finite impulse response filters respectively, in accordance with said first and second impulse response characteristics respectively;

first and second controllable delay means coupled to receive output signals produced from said first and second finite impulse response filters respectively; and,

delay time setting means for adjusting respective amounts of delay produced by said first and second controllable delay means such as to compensate for differences in respective signal transfer delay times of said first and second finite impulse response filters.

26. A digital equalizer apparatus according to claim 25, in which said amplitude/frequency characteristic is supplied from said input means as a set of amplitude values corresponding to respective sample frequencies within a predetermined frequency range, and in which the number of said amplitude values supplied to said transform operational means when said Hilbert phase transform operation is selected is twice the number of said amplitude values supplied when said linear phase transform operation is selected.

27. A digital equalizer apparatus according to claim 25, and further comprising interpolation means, in which said amplitude/frequency characteristic is supplied from said input means as a set of amplitude values corresponding to respective sample frequencies within a predetermined frequency range and in which when said Hilbert transform operation is selected, interpolation of said amplitude values from said input means is executed and amplitude values thereby derived are combined with said amplitude values from said input means to thereby double the number of amplitude values supplied to said transform operational means for computing said transfer functions.

28. A digital equalizer apparatus comprising:

finite impulse response filter means;

impulse signal generating means for generating an impulse test signal;

signal output means;

signal input means for transferring an input signal to said finite impulse response filter means as a train of digital samples;

selector means operable for selectively transferring an output signal from said filter means and said impulse test signal to said signal output means to be supplied to an external system, and selectively transferring an input audio signal and a measurement signal produced by applying said test signal to said external system to be inputted to said signal input means;

amplitude input means for inputting data representing an arbitrary amplitude/frequency characteristic;

phase input means for inputting data representing an arbitrary phase/frequency characteristic;

input transfer function operational means for computing a transfer function corresponding to said input amplitude/frequency characteristic and phase/frequency characteristic;

memory means for storing an impulse response waveform of said measurement signal, supplied from said signal input means;

Fourier transform means for deriving the Fourier transform of said impulse response waveform stored in said memory means;

conjugate transfer function operation means for computing the conjugate transfer function of said transfer function derived by said Fourier transform means;

inverse amplitude transfer function operational means for deriving a transfer function in which each amplitude value at each frequency within a frequency range of said transfer function is the inverse of an amplitude value of said transfer function derived by said conjugate transfer function operational means, at that frequency, and in which each phase value at each frequency in said range is unchanged from a phase value of said transfer function derived by said conjugate transfer function operational means, at that frequency;

convolution means for computing the convolution of said transfer function derived by said input transfer function operational means and said transfer function derived by said inverse amplitude transfer function operational means, to derive a set of filter coefficient values in accordance with an impulse response characteristic obtained as a result of said convolution operation; and,

setting means for setting into said finite impulse response filter means respective filter coefficients having said coefficient values.

29. A digital equalizer apparatus according to claim 28, in which said convolution means derives the inverse Fourier transform of a transfer function that is derived as the convolution in the time domain of said transfer function obtained by said inverse amplitude transfer function operational means and said transfer function obtained by said input transfer function operational means.

30. A digital equalizer apparatus according to claim 28, in which said convolution means derives the inverse Fourier transforms of both said transfer function that is obtained by said inverse amplitude transfer function operational means and said transfer function obtained by said input transfer function operational means, and derives the convolution of the real components of the resultant inverse Fourier transforms in the time domain.

31. A digital equalizer apparatus according to claim 28, and further comprising window function means for multiplying data stored in said memory means by a predetermined window function, with the Fourier transform of a resultant output from said window function means being derived by said Fourier transform means.

BACKGROUND OF THE INVENTION

The present invention relates to a digital equalizer apparatus incorporating a digital filter for frequency compensation of an audio signal which has been converted to digital code sample form. In particular, the invention relates to such an apparatus which employs a FIR (finite impulse response) digital filter, and enables mutually independent adjustment of the amplitude/frequency and phase/frequency response characteristics of the filter.

With the equivalent of audio apparatus utilizing digital signals in recent years, digital equalizers have been developed based upon FIR filters. In the following, it will be assumed that a FIR filter is a transversal filter, i.e. a tapped delay line filter. However it should be noted that the present invention is not limited to such an FIR filter, and that other filter configurations can be utilized. The transfer function of such a digital transversal filter, determined by the amplitude/frequency characteristic and phase/frequency characteristic of the filter, is determined by the respective values of a plurality of filter coefficients (sometimes referred to as tap coefficients). Such an FIR filter has been utilized in the prior art for audio digital equalizer. However in the prior art it has not been possible to execute mutually independent control of the phase and amplitude response characteristics of such an audio equalizer by using a single FIR filter, i.e. for thereby independently modifying the amplitude/frequency characteristic and phase/frequency characteristic of a digital audio signal by transferring the signal through the FIR filter.

In addition to such audio equalizer applications, a digital equalizer apparatus based on a FIR filter can be adapted to various other functions, for example suppression of "howl" caused by acoustic feedback between a microphone and a loudspeaker.

FIG. 1 is a system block diagram of an example of a prior art digital equalizer apparatus based on a FIR filter. Numeral 1 denotes an amplitude/frequency characteristic input section, for input of data which represent an arbitrary amplitude/frequency characteristic that will be designated as .vertline.H(.omega.).vertline.. Numeral 5 denotes an inverse Fourier transform section which operates on the input amplitude/frequency characteristic as a transfer function, and derives the inverse Fourier transform of this transfer function. This inverse Fourier transform is an impulse response characteristic corresponding to the transfer function, as described hereinafter, and a set of values of filter coefficients respectively determined by that impulse response characteristic is thereby obtained. Numeral 6 denotes setting means for establishing these values of filter coefficients for a FIR filter 7, to thereby determine the desired amplitude/frequency characteristic for the filter. Numeral 8 denotes a signal input section for converting an input signal to suitable digital signal form to be processed by the FIR filter 7, and 9 denotes a signal output section for converting a digital output signal produced from the FIR filter 7 to a suitable form for transfer to external circuits.

Data representing the desired amplitude/frequency characteristic .vertline.H(.omega.).vertline. are inputted through the amplitude/frequency characteristic data input section 1, as a set of amplitude values corresponding to respective frequencies, referred to in the following as sample frequencies. FIG. 2(A) shows an example of such an amplitude/frequency characteristic, in which these input amplitude values are indicated as black dots, with data being inputted only within a frequency range designated as 0 to .pi.. As shown in FIG. 2(B), the desired amplitude/frequency characteristic in the range 0 to 2 can be derived by "folding over" the portion of the characteristic from 0 to .pi. and thereby obtaining the characteristic in the range .pi. to 2.pi..

The amplitude/frequency characteristic in the range 0 to 2 thus obtained is applied to the inverse Fourier transform section 5, where the inverse Fourier transform is derived. More specifically, the amplitude/frequency characteristic .vertline.H(.omega.).vertline. is treated as if it were the absolute amplitude portion of a transfer function H(.omega.), i.e.

H(.omega.)=.vertline.H(.omega.).vertline. (1)

As is well known, the inverse Fourier transform of a transfer function (which is a complex function in the frequency domain) is a time domain function which represents the impulse response of the circuit having that transfer function. Thus, the inverse Fourier transform of the transfer function H(.omega.) is derived by the inverse Fourier transform section 5, to thereby obtain a desired impulse response for the FIR filter 7 corresponding to the input amplitude/frequency characteristic from input section 1. Since the respective values of filter coefficients of a transversal filter are inherently defined by corresponding values of the impulse response of the filter, the appropriate filter coefficient values for the FIR filter 7 are thereby determined. These values are then set in the FIR filter 7 by the setting section 6 (e.g. by control signals applied from section 6), so that the amplitude/frequency characteristic of the FIR filter 7 is thereby made identical to that inputted from input section 1.

The inverse Fourier transform is executed in accordance with the following equation:

h(n)=1/N.times..SIGMA.H(.omega.).times.e.sup.j.omega. h (2)

In the above, .omega.=2.times..pi./N.times.k 0.ltoreq.n.ltoreq.(N-1)

The values h(n) obtained from equation (2) are the filter coefficients that are established for the FIR filter 7 by the setting section 6. The FIR filter 7 thereby realizes the specified amplitude/frequency characteristic. However the phase/frequency characteristic of the FIR filter 7 is determined by the transfer function of equation (1) above, and so is fixed as an inherently linear characteristic.

Thus with the prior art example of FIG. 1, although it is possible to realize an arbitrary shape of amplitude/frequency characteristic for the FIR filter 7, the phase/frequency characteristic of the filter is inherently defined by the filter coefficients to be linear. It is thus a disadvantage of such a prior art apparatus that it is not possible to mutually independently establish an arbitrary shape of phase/frequency characteristic and an arbitrary shape of amplitude/frequency characteristic, using a single FIR filter.

In addition to the above, problems also arise even if an equalizer apparatus is implemented which is capable of being adjusted to produce such arbitrary phase and amplitude responses (e.g. by using separate FIR filters for these responses). For example if it is desired that the FIR filter will realize the amplitude/frequency characteristic and phase/frequency characteristic of a specific circuit or system, then it is necessary to first measure that amplitude/frequency characteristic and phase/frequency characteristic of the circuit or system and to then input measured data representing the amplitude/frequency characteristic and the phase/frequency characteristic respectively to respective amplitude and phase input means. Moreover if it is desired to realize, using such a FIR filter apparatus, an amplitude/frequency characteristic and phase/frequency characteristic that have been computed, then there is no simple way of inputting that amplitude/frequency characteristic and phase/frequency characteristic for establishing the desired FIR filter response.

SUMMARY OF THE INVENTION

It is an objective of the present invention to provide a digital equalizer apparatus utilizing a FIR filter, whereby an arbitrary amplitude/frequency characteristic and an arbitrary phase/frequency characteristic for the filter can be established mutually independently.

It is a further objective of the present invention to provide a digital equalizer apparatus utilizing a FIR filter, whereby the amplitude/frequency characteristic and phase/frequency characteristic of the FIR filter can be easily modified to achieve compensation for frequency response characteristics of one or more components of an audio system.

It is a further objective of the present invention to provide a digital equalizer apparatus utilizing a FIR filter, whereby data representing a desired amplitude/frequency characteristic and phase/frequency characteristic for the FIR filter can be inputted to the digital equalizer apparatus in the form of parameters of a specific circuit having an amplitude/frequency characteristic and phase/frequency characteristic each of which is controlled by these parameters in a known manner, such as resonance-related parameters of a circuit exhibiting resonance at a single frequency.

It is a further objective of the present invention to provide a digital equalizer apparatus whereby an improved degree of frequency resolution for equalization is achieved over a frequency range extending down to substantially low values of frequency, while maintaining a high level of processing speed for operation of a FIR filter within the digital equalizer apparatus.

To achieve the above objectives, a digital equalizer apparatus according to the present invention comprises:

amplitude/frequency input means for inputting amplitude/frequency characteristic data representing an arbitrary amplitude/frequency characteristic;

phase data input means for inputting data for establishing a phase/frequency characteristic, said data being in a category selected from a group of categories of data which includes phase/frequency characteristic data group delay characteristic data, amplitude/frequency characteristic data, and data expressing a resonance condition of a predetermined type of electrical circuit;

phase/frequency operational means for computing phase/frequency characteristic data based upon said data from said phase data input means;

transfer function operational means for operating on said amplitude/frequency characteristic and phase/frequency characteristic data to derive transfer function data representing a transfer function;

inverse Fourier transform means for operating on said transfer function data to derive impulse response characteristic data representing an impulse response characteristic determined by said transfer function;

finite impulse response filter means

signal input means for transferring to said finite impulse response filter means an input audio signal as a train of digital samples;

signal output means for receiving said audio signal after frequency characteristic modification of said audio signal by said finite impulse response filter means, and for transferring the modified audio signal to an external system; and,

setting means operable for establishing a set of filter coefficients for said finite impulse response filter means having respective values determined by said impulse response characteristic.

In another aspect, with a digital equalizer apparatus according to the present invention as set out above, said data from said phase data input means represent a group delay characteristic, and said phase/frequency operational means comprises integrator means for integrating said group delay characteristic data with respect to frequency, for thereby deriving said phase/frequency characteristic to be supplied to said transfer function operational means.

In another aspect, with a digital equalizer apparatus according to the present invention as set out above, said finite impulse response filter means comprises a plurality of finite impulse response filters, and the apparatus further comprises:

a plurality of digital band-pass filters for dividing said digital sample signal from said signal input means into a plurality of frequency bands:

a plurality of down-sampling sections for receiving respective band-divided output signals from said band-pass filters for reducing the sampling frequency of said band-divided output signals by respectively differing reduction factors, and supplying resultant digital sample signals to respective ones of said plurality of finite impulse response filters.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a system block diagram of a prior art digital equalizer utilizing a FIR filter;

FIGS. 2(A) and 2(B) are diagrams for illustrating input of amplitude/frequency characteristic data to a digital equalizer and derivation of extended amplitude/frequency characteristic data therefrom;

FIG. 3(A) is a system block diagram of a first embodiment of a digital equalizer according to the present invention, in which group delay characteristic data are used to define a phase/frequency characteristic;

FIG. 3(B) is a flow chart for use in describing the operation of the first embodiment;

FIG. 4 is a system block diagram of a second embodiment of a digital equalizer according to the present invention, in which input group delay characteristic data are redefined with respect to an average group delay, to derive a phase/frequency characteristic;

FIG. 5 is a system block diagram of a third embodiment of a digital equalizer according to the present invention, in which resonance data for a low pass filter circuit are inputted to define a phase/frequency characteristic;

FIGS. 6(A) and 6(B) are circuit diagrams of examples of second order active low pass filter circuits;

FIG. 6(C) shows amplitude/frequency characteristic and phase/frequency characteristic examples for a second order low pass filter;

FIG. 7(A) is a system block diagram of a fourth embodiment of the present invention, in which resonance data are inputted to define a phase/frequency characteristic;

FIGS. 7(B) and 7(C) show phase/frequency characteristics for assistance in describing the operation of the fourth embodiment;

FIGS. 8 and 9 are system block diagrams of fifth and sixth embodiments of the present invention, in which amplitude/frequency characteristic data are inputted as phase data, for deriving a phase/frequency characteristic by Hilbert transform computation;

FIG. 10(A) is a system block diagram of a seventh embodiment of the present invention, in which different frequency bands of a digital audio signal are subjected to down-sampling and are processed in parallel by a plurality of FIR filter channels;

FIG. 10(B) is a flow chart for describing the operation of the seventh embodiment;

FIGS. 11 and 12 are system block diagrams of eighth and ninth embodiments of the invention in which different frequency bands of a digital audio signal are subjected to down-sampling and are processed in parallel by a plurality of FIR filter channels;

FIG. 13 is a system block diagram of a tenth embodiment of the present invention, enabling equalization for the acoustic characteristics of a sound field;

FIG. 14 is a diagram for illustrating the derivation of an amplitude deviation/frequency characteristic of the tenth embodiment;

FIGS. 15(A) to (D) are amplitude/frequency characteristic and phase/frequency characteristic diagrams for describing the operation of the tenth embodiment;

FIG. 16(A) is a system block diagram of an 11th embodiment of the present invention, which enables generation of test signals to drive a loudspeaker and sound field, and analysis of the resultant frequency response for executing equalization;

FIG. 16(B) is a flow chart for describing the operation of the 11th embodiment;

FIGS. 17, 18, 20, 22, 24, 27 and 28 are system block diagrams of 12th, 13th, 14th, 15th and 16th embodiments of the present invention respectively, enabling equalization for the acoustic characteristics of a sound field by analyzing a stored measurement signal waveform and modifying a FIR filter phase/frequency characteristic and amplitude/frequency characteristic accordingly;

FIGS. 19 (A) and (B) are diagrams for illustrating a signal level decision operation of the 13th embodiment;

FIGS. 21(A) to (C) are diagrams for illustrating removal of uncorrelated noise components of a measurement signal in the 14th embodiment;

FIGS. 23(A) and (B) are diagrams for illustrating sampling of an initial portion of a measurement signal of the 15th embodiment;

FIG. 23 (C) is a flow chart for describing the operation of the 15th embodiment;

FIGS. 25 (A) to (C) and 26(A) to (F) are diagrams for assistance in describing window function operations executed by the 16th embodiment;

FIG. 29 is a system block diagram of a 19th embodiment of the present invention, with a memory having stored therein phase/frequency characteristics for use in compensating a loudspeaker group delay characteristic in a plurality of frequency ranges, together with related amplitude/frequency characteristic data, which are applied to modify a FIR filter transfer function;

FIGS. 30(A) to (C) are diagrams for illustrating group delay characteristic compensation by the 19th embodiment;

FIG. 31 is a system block diagram of a 20th embodiment of the present invention, in which an inverse Fourier transform of a transform function computed for a FIR filter is multiplied by a window function before utilization for establishing filter coefficients;

FIG. 32 is a system block diagram of a 21st embodiment of the present invention, enabling equalization for the phase/frequency characteristic of a loudspeaker;

FIG. 33 is an equivalent circuit diagram of a loudspeaker;

FIG. 34 is a partial system block diagram of a 22nd embodiment of the present invention, whereby either a linear transform method or Hilbert transform method can be selected for computing filter coefficients;

FIG. 35 is a flow chart for describing the operation of the 22nd embodiment;

FIG. 36 is a block diagram illustrating switch selection of input data for the 22nd embodiment;

FIG. 37 is a flow chart for describing a 23rd embodiment of the present invention;

FIGS. 38(A) and (B) are characteristic diagrams for describing an interpolation operation of the 23rd embodiment;

FIG. 39 is a system block diagram of a 24th embodiment of the present invention, enabling the acoustic characteristics of a sound field to be analyzed and corresponding equalization implemented;

FIG. 40 is a flow chart for describing the operation of the 24th embodiment;

FIGS. 41(A) and (B