JPH073935B2 - Sound quality adjustment device - Google Patents

Description

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a sound quality adjustment apparatus using a transversal filter (hereinafter, FIR filter) that realizes an arbitrary frequency characteristic.

2. Description of the Related Art FIG. 4 shows a block diagram of a conventional sound quality adjusting apparatus. In the figure, 1 is an input means for inputting an arbitrary amplitude frequency characteristic for each sampling frequency, and 2 is an input amplitude frequency characteristic | H
(Ω) | is used to obtain a transfer function, and a filter coefficient is calculated from the transfer function by an inverse Fourier transform. A filter coefficient calculating means 3 is a FIR filter 5a that realizes the amplitude frequency characteristic of the obtained filter coefficient. 5b is a coefficient setting means, 4 is a delay device 11 and low-pass filters 12, 13
A band dividing unit configured to divide an actual digital signal into a plurality of bands, 6 is down-sampling means for lowering the sampling frequency by thinning out the band-divided signals, and 7 is an up-converting sampling frequency. Sampling means, 8 is a low-pass filter for removing high-frequency noise generated by upsampling, 9 is a multiplier for matching the gains of the low frequency band and the high frequency band, and 10 is a time difference between the low frequency band and the high frequency band generated by calculation. It is a delay device for correction.

The operation of the sound quality adjusting device configured as described above will be described below.

The desired amplitude frequency characteristic | H (ω) | is input by the input means 1 for each sampling frequency ω = 2π / Nxk (k = 0 to N−1, N: the number of sampling points). Next, the transfer function H (ω) is obtained by the filter coefficient calculation means 2.

The desired frequency characteristic to be realized usually does not include phase information because it is represented by the amplitude frequency characteristic (power spectrum). Therefore, as a method for obtaining phase information from the power spectrum, two methods using the Hilbert transform and the relationship between linear phases are used.

The obtained transfer function is inverse Fourier transformed into an impulse response, and the coefficient setting means 3 causes the FIR filters 5a and 5b.
Is set to.

Further, the delay device 10 is set with a delay time for correcting the time difference between the low band and the high band caused by the band division.

The input amplitude frequency can be realized as described above.

DISCLOSURE OF THE INVENTION Problems to be Solved by the Invention However, in the above-described conventional configuration, the cutoff frequency of the low-pass filter for band division and the low sampling frequency are fixed even if the frequency resolution of a certain band is made fine as necessary. However, there was a drawback that it could not be realized.

In view of the above problems, the present invention can change the frequency of a low-pass filter for dividing the low band and the high band and the sampling frequency of the low band, and if necessary, the frequency of the necessary band (particularly the low band). An object of the present invention is to provide a sound quality adjustment device capable of fine resolution.

Means for Solving the Problems In order to solve the above problems, the present invention has a low-pass filter, which divides an input digital signal into a plurality of bands and a band-divided signal. A transversal filter for performing a convolution process, a delay device for time-matching the signals for each band subjected to the convolution process, an input unit for inputting an arbitrary frequency characteristic, and band-dividing the input frequency characteristic, A filter coefficient calculating means for obtaining an impulse response by inverse Fourier transforming a transfer function for each band and a real part of the obtained impulse response are set as a filter coefficient in a transversal filter,
Further, it comprises a coefficient setting means for setting the coefficient of the low-pass filter of the band division part, and down-sampling means for lowering the sampling frequency by thinning out the signals band-divided by the band division part. The division cutoff frequency and the sampling frequency of the digital signal band-divided by the band division unit are variable.

The present invention has the above-mentioned configuration, by changing the cutoff frequency of the band division, obtaining the impulse response for each band, changing the cutoff frequency of the low-pass filter of the band division unit, and changing the thinning interval by the downsampling means to change the low frequency range. By changing the sampling frequency of, it becomes possible to make the frequency resolution of the low frequency band fine.

Embodiment One embodiment of the present invention will be described below with reference to the drawings. FIG. 1 is a block diagram of a filter coefficient calculation device according to an embodiment of the present invention.

In FIG. 1, 1 is an input means for inputting the input amplitude frequency characteristic for each sampling frequency, and 2 is a filter coefficient calculation for obtaining an impulse response by dividing the input amplitude frequency characteristic into bands and calculating a transfer function of each band. Means 3 uses the real part of the impulse response as a filter coefficient, and FIR filters 5a and 5b.
And coefficient setting means 4 for setting the coefficients of the low-pass filters 8, 12 and 13 are a delay device 11 and a low-pass filter 1.
A band dividing unit configured by 2, 13 for band-dividing an actually input digital signal, 6 is down-sampling means for lowering the sampling frequency by thinning out the band-divided signal, and 5a, 5b are obtained as described above. FIR filter for performing an operation according to the filter coefficient, 7 is up-sampling means for returning the sampling frequency based on, 8 is a low-pass filter for removing high-pass noise generated by up-sampling, and 9 is a low-pass and high-pass gain. Multiplier for matching, 10
Is a delay device that corrects the time lag between the low band and the high band caused by the calculation.

The operation of this embodiment will be described below with reference to the drawings.

In the input means 1, a desired amplitude frequency characteristic | H (ω) | is input for each sampling frequency. However, particularly when it is desired to make the frequency resolution of the low frequency band fine, the low frequency band is input with a fine resolution. For example, if you want to change the low frequency decomposition farm from 20Hz to 10Hz, input only every low frequency in 10Hz.

Next, the filter coefficient calculation means 2 reduces the cutoff frequency of the band division to about 1/2, divides the band into the low band and the high band, and calculates the transfer function for each band. The obtained transfer function is subjected to inverse Fourier transform and becomes an impulse response, which is set in the high frequency FIR filter and the low frequency FIR filter of the FIR filters 5a and 5b.

In the band division unit 4, the cutoff frequency of the low-pass filter 11 for band division is reduced to about 1/2 and filtering is performed.

Next, the down-sampling means 6 doubles the intervals of the thinned signals, and the FIR filter 5b performs the convolution processing. 2a and 2b show the method of downsampling. The convolved signal is upsampling means 7
Inserts a zero value at the original sampling frequency and restores the sampling frequency. The low pass filter 8 removes high frequency noise generated by the upsampling process, and the multiplier 9 adjusts the level. On the other hand, the high frequency signal is received by the FIR filter 5
After being convolved with a, the delay unit 10 time-matches the low-frequency signal and adds it to the low-frequency signal. FIGS. 3a and 3b show the frequency characteristics realized by the present invention.

With the above method, it is possible to make the frequency resolution in the low frequency band fine as necessary.

Although the band division of the amplitude frequency characteristic is divided into two in the present embodiment, the same effect can be obtained by any number of divisions.

Further, in this embodiment, the resolution in the low frequency range is doubled, but n of 2 is used.
It may be a power (n is an integer) times.

Advantageous Effects of InventionThe present invention determines the impulse response of each band by changing the cutoff frequency of the band division according to the resolution and the input means for inputting the amplitude frequency characteristic, and sets the real part of the impulse response as a filter coefficient in the FIR filter. In addition, by setting the coefficient in the low-pass filter of the band division unit and changing the downsampling interval, the frequency resolution of the required band can be set in the FIR filter with a finer resolution, and the desired amplitude frequency characteristics can be realized accurately. Therefore, it is possible to provide a sound quality adjusting device.

[Brief description of drawings]

FIG. 1 is a block diagram of a sound quality adjusting device in an embodiment of the present invention, FIG. 2 is an amplitude setting diagram showing a downsampling method, FIG. 3 is a characteristic diagram of the same, and FIG. 4 is a conventional sound quality adjusting device. It is a block diagram of. 1 ... Input means, 2 ... Filter coefficient calculation means, 3 ...
Setting means, 4 ... band division unit, 5a, 5b ... FIR filter,
6 ... Down sampling means, 7 ... Up sampling means, 8 ... Low pass filter, 10 ... Delay device.

Claims (1)

[Claims] 1. A band division unit having a low-pass filter, which divides an input digital signal into a plurality of bands, a transversal filter which performs convolution processing on each band-divided signal, and the convolution processed band. A delay device for adjusting the time of each signal, input means for inputting arbitrary frequency characteristics, band division of the input frequency characteristics, obtaining a transfer function for each band, inverse Fourier transform and impulse response And a coefficient setting unit that sets the real part of the obtained impulse response to the transversal filter as a filter coefficient, and sets the coefficient of the low-pass filter of the band division unit,
Down-sampling means for lowering the sampling frequency by thinning out the band-divided signal in the band-dividing unit, the band-dividing cut-off frequency of the band-dividing unit and the digital signal band-divided in the band-dividing unit. A sound quality adjusting device having a variable sampling frequency.