JPH0720309B2 - Sound quality adjustment device for speakers - Google Patents

Description

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a speaker sound quality adjusting device for processing a music signal input to a speaker system through a power amplifier in the hall acoustic field.

2. Description of the Related Art In recent years, there has been a remarkable expansion of digital signal processing technology into the acoustic field, and particularly in the hall acoustic field, the development of digital mixers, digital equalizers, and the like has enabled more detailed sound production. In addition, a dedicated channel divider is used in the speaker system, and simple unit correction is performed in addition to band division. Together with the digital signal processing technology, sound quality adjustment technology in the hall sound system is making remarkable progress. is there.

An example of the above-described conventional hall sound system will be described below with reference to the drawings.

FIG. 7 is a system diagram of a hall acoustic system in the conventional example. In FIG. 7, 21 is a microphone and 22 is a mixer. 23 is an equalizer, 39
Is a channel divider. 25 is a power amplifier for driving, 26 is a speaker system, and 27, 28, and 29 are tweeters for high-range reproduction, midrange for middle-range reproduction, and woofer for low-range reproduction, respectively.

The operation of the hall sound system system diagram configured as described above will be described below.

First, the music signal from the microphone 21 is mixed by the mixer 22 and input to the equalizer 23. The equalizer 23 performs optimum filtering on the input music signal in consideration of the characteristics of the hall or the listening room or the characteristics of each unit attached to the speaker system 26, and inputs the music signal to the channel divider 39.
The channel divider 39 band-divides the input music signal and outputs it as a music signal for each unit. In some cases, a filtering function for correcting the frequency characteristic of each unit and a delay function for phase matching may be provided. The output signal from the channel divider 39 is driven by the power amplifier 25 and input to each speaker unit such as the tweeter 27, the midrange 28, and the woofer 29 of the speaker system 26.

Problems to be Solved by the Invention However, in the configuration as described above, the filter incorporated in the channel divider 39 has a tone adjusting function constituted by analog elements, and the equalizer 23 has a 1/3 ratio. Even if an octave band is used, it is difficult to change the sound pressure level of only one band independently and suddenly. There was a problem that the above steep peak and dip could not be completely flattened. Moreover, even if a highly accurate digital equalizer is used as the equalizer 23, there is a problem that it takes time to set.

In view of the above problems, the present invention flattens abrupt peaks and dips on the sound pressure-frequency characteristics of each speaker unit used in a speaker system, and makes it possible to easily and quickly provide an acoustic system in a concert venue such as a hall. The present invention provides an excellent speaker sound quality adjustment device that can be set.

Means for Solving the Problems In order to solve the above problems, a speaker sound quality adjusting apparatus according to the present invention includes a first low-pass filter for low-pass filtering a music signal input from an input terminal, and a low-pass filter. An analog-digital converter for converting the filtered music signal into a digital signal, a first digital filter for high-pass filtering the digital signal and correcting the frequency characteristic of the high-pitched speaker unit, and the digital signal A second digital filter for band-filtering the sound signal and correcting the frequency characteristic of the speaker unit for middle sound, and a third digital filter for low-pass filtering the digital signal and correcting the frequency characteristic of the speaker unit for low sound. The output of the filter and the first, second, and third digital filters are analog. A digital-analog conversion unit for converting the analog signals into low-pass filters, second, third, and fourth low-pass filters for low-pass filtering each of the unlogged signals, an arithmetic unit, and an interface unit. A unit characteristic storing means for storing impulse response of each of the high-pitched speaker unit, the middle-tone speaker unit, and the low-pitched speaker unit, a target characteristic input unit for inputting a target amplitude frequency characteristic to be realized, and the target amplitude frequency. Hilbert transform means for calculating the displacement characteristics by Hilbert transform of the characteristics, inverse Fourier changing means for converting the amplitude and phase characteristics output from the Hilbert transform means into time axis characteristics, the time axis characteristics and the unit characteristic memory The tap coefficient of each digital filter is calculated from the impulse response output from the means. Consists of a tap coefficient calculating means, the calculated first tap coefficients via said interface portions, second, and is characterized in that the forwarding to the third digital filter.

According to the present invention, the unit characteristics of each speaker unit stored in the unit characteristic storage unit and the target amplitude characteristic to be realized input by the target characteristic input unit are subjected to Hilbert conversion by the Hilbert conversion unit and phased. Based on the target transmission characteristic obtained by calculating the characteristic, the tap coefficient calculating means calculates the tap coefficient that cancels the characteristic peculiar to the unit and realizes the target transmission characteristic. The tap coefficient is transferred to each digital filter through the interface unit. The music signal input from the input terminal is low-pass filtered by a low-pass filter, converted into a digital signal by an analog-digital conversion unit, and input to each digital filter. Each digital filter performs band-pass filtering suitable for each speaker unit, flattens the steep peak dip on the frequency characteristics of each speaker unit, and performs filtering for realizing a target frequency characteristic with high accuracy. The output from each digital filter is converted into an analog signal by a digital-analog converter, and then low-pass filtered by a low-pass filter and output. By the above operation, it is possible to completely eliminate the sharp peak dip in the sound pressure frequency characteristic due to each speaker unit that causes a hearing problem with simple and single-time operation, and realize the target frequency characteristic with high accuracy. Becomes

Embodiment Hereinafter, a speaker sound quality adjusting apparatus according to an embodiment of the present invention will be described with reference to the drawings. FIG. 1 is a block diagram of a speaker sound quality adjusting apparatus according to a first embodiment of the present invention. In FIG. 1, 1 is an input terminal,
2,10,11,12 are low pass filters (hereinafter referred to as LPF),
3 is an analog-digital conversion unit (hereinafter referred to as A / D conversion unit), 4, 5 and 6 are digital filters, 7, 8 and 9 are digital-analog conversion units (hereinafter referred to as D / A conversion unit), and 13 is a tweeter terminal , 14 is a midrange terminal, 15 is a woofer terminal, and 16 is an interface for connecting to a computer 17.

In this embodiment, the unit characteristic storage means, which is a component of the present invention described in the claims,
The target characteristic input means, the Hilbert transform means, the inverse Fourier transform means, and the tap coefficient calculation means will be described in detail later.
It is embodied by the software of the computer 17 and peripheral devices.

The operation of the speaker sound quality adjusting apparatus configured as described above will be described below with reference to FIGS. 1, 2, 3, 4, and 5.

First, FIG. 2 is a system diagram of a hall sound system including the present invention. A music signal input from a microphone 21 passes through a mixer 22, an equalizer 23 for room sound field correction such as a hall, and a sound quality for a speaker. It is input to the adjusting device 24. In FIG. 1, the input music signal is low-pass filtered by LPF2 and then converted into a digital signal in the A / D converter 3. Digital signals are input in parallel to a plurality of digital filters 4, 5 and 6, and tweeters 27, midrange 28 and woofer 29 of each speaker unit shown in FIG.
The speaker unit is divided into the frequency bands that the speaker unit handles, and the peaks and dips on the sound pressure frequency characteristics of each speaker unit are flattened and subjected to filtering to obtain the frequency characteristics desired by the operator. The tap coefficient to be given to each digital filter 4, 5 and 6 in order to realize such filtering is transferred to the memory of each digital filter 4, 5 and 6 through the interface 16 after being calculated in advance by the computer. However, the algorithm will be described later. The digital signals output from the digital filters 4, 5 and 6 are D / A converters 7, 8 and
After being demodulated by 9, it is low-pass filtered by low-pass filters 10, 11, 12 and output from each output terminal 13-15. In FIG. 2, the music signal output from the speaker sound quality adjusting device is amplified by the power amplifier 25 and drives each speaker unit constituting the speaker system 26.

Next, the procedure and algorithm for calculating the tap coefficient to be transferred to each digital filter 4, 5, 6 will be described.

FIG. 3 is a diagram showing a digital filter tap coefficient calculation algorithm. First, in FIG. 2, the target frequency characteristic is input using the light pen 20 and the display 19 as the target characteristic inputting means (step 30). The input target frequency characteristic is amplitude information or real part information, and a Hilbert transform is applied to this to obtain a transmission characteristic, which is divided for each band that each speaker unit handles (step 31). Step 31 corresponds to the Hilbert transforming means which is a constituent element of the present invention described in the claims. Further, the inverse Fourier transform is applied to each band to obtain the impulse response of the target characteristic for each band (step 32). Step 32 corresponds to the inverse Fourier transform means which is a component of the present invention described in the claims. Next, in FIG. 2, the impulse response of each speaker unit recorded in the external storage device 18 as the unit characteristic storage means is read (step 3
3) Obtain the inverse characteristics, and perform convolution integration of these inverse characteristics and the target characteristics divided into respective corresponding bands (step 34) to calculate the tap coefficient.
Step 34 corresponds to the tap coefficient calculating means which is a constituent element of the present invention described in the claims. Then, the calculated tap coefficient is output from the computer 17 through the interface 16 to the memory of each digital filter 4, 5, 6 (step 35).

As a specific example, FIG. 4 shows how the mid-range digital filter tap coefficient is calculated when the target frequency characteristic is flat within the reproduction band of the speaker system.
It is shown in FIG. FIG. 4 is a graph showing the target frequency characteristic of the speaker system, which is input from the display 19 using the light pen 20 in FIG. In FIG. 4, the target frequency characteristic is also shown with the preset cutoff frequencies F1 and F2 and the characteristic obtained by band division by the cutoff characteristic. FIG. 5 (a) shows the anechoic chamber characteristics of the midrange unit, and is stored in the external storage device 18 of FIG. 2 as the impulse response including the phase information. FIG. 5 (b) shows the tap coefficients transferred to the mid-range digital filter 5 on the frequency axis, and shows that the mid-range target characteristics shown in FIG. 4 have been subjected to Hilbert transform and inverse Fourier transform. The mid-range unit shown in FIG. 5 (a) is obtained by convolution integration with an anechoic chamber characteristic which is an impulse response. FIG. 5 (c) shows the mid-range anechoic chamber characteristic corrected by the digital filter 5 using the tap coefficient thus obtained. In FIGS. 4 and 5, F1 and F2 are cutoff frequencies, and FS 'is a sampling frequency (Nyquist).

As described above, according to this embodiment, the music signal low-pass filtered by the LPF 2 is converted into a digital signal by the A / D converter 3, and the digital signal is converted into a plurality of digital filters 4 to 6 arranged in parallel. To the computer 17 connected to the external storage device 18, the light pen 20, and the input device such as the display through the interface 16 to control the filter characteristics of the digital filters 4 to 6. By
Simply by performing the simple operation of inputting the target frequency characteristic, it is possible to flatten the peak and dip of the sound pressure frequency characteristic caused by each speaker unit that is a problem in hearing, and to quickly create a sound that matches the indoor environment. Can be done.

A second embodiment of the present invention will be described below with reference to the drawings.

FIG. 6 is a block diagram of a speaker sound quality adjusting apparatus according to a second embodiment of the present invention. In the figure, 1 is an input terminal, 2 is an A / D converter, 3,10,11,12 are LPFs, 4,5,6 are digital filters, 7,8,9 are D / A converters, and 13 is Tweeter terminal, 14 midrange terminal, 15 woofer terminal,
The above is the same as the configuration of FIG. The difference from the configuration shown in Fig. 1 is that instead of connecting each digital filter to a computer through an interface, ROM36,3
This is the point where 7,38 are connected.

In the speaker sound quality adjusting device configured as described above, the tap coefficients of the respective digital filters transferred from the computer through the interface are stored in the ROMs 36, 37 and 38. Therefore, when there is no change in the speaker system used in the field of equipment acoustics, etc., the tap coefficient calculated once is stored in ROM and the tap coefficient is accessed from this ROM. The number of devices can be reduced and the setting of each audio device can be completed in a shorter time.

In addition, in the first and second embodiments, the midrange 28 that handles the midrange is one unit, but the midrange 28 is further divided into two units.
4 with digital filter 5, D / A converter 8 and low-pass filter 11
The system may be more than a way.

EFFECTS OF THE INVENTION As described above, the present invention provides first, second, and third corrections for correcting the frequency characteristics of high-pitched, mid-tone, and low-pitched speaker units.
A third digital filter, a digital-analog converter for converting the outputs of the first, second, and third digital filters into analog signals, and a second analog filter for low-pass filtering the converted analog signals. A third and a fourth low pass filter, an arithmetic unit and an interface unit,
The calculation unit includes a unit characteristic storage unit that stores impulse responses of the high-pitched speaker unit, the middle-tone speaker unit, and the low-pitched speaker unit, and a target characteristic input unit that inputs a target amplitude frequency characteristic to be realized. Hilbert transform means for calculating a phase characteristic by Hilbert transforming the target amplitude frequency characteristic, inverse Fourier transform means for transforming the amplitude and phase characteristic output from the Hilbert transform means into a time axis characteristic, and the time axis characteristic And a tap coefficient calculation means for calculating a tap coefficient of each digital filter from the impulse response output from the unit characteristic storage means, and the calculated tap coefficient is passed through the interface section to the first and second , The third digital filter is used for transfer, It is possible to completely flatten the steep peak and dip in the sound pressure frequency characteristics due to the problematic speaker units, and to easily and quickly obtain the frequency characteristics desired by the user in a concert venue or on the spot. It will be possible to adjust the sound system and adjust the sound quality.

[Brief description of drawings]

FIG. 1 is a block diagram of a sound quality device for a speaker unit according to a first embodiment of the present invention, FIG. 2 is a hall sound system system diagram when the present invention is used, and FIG. 3 is a digital filter tap coefficient calculation algorithm. Shown, 4th
The figure shows the target frequency characteristic curve diagram of the speaker system input by the operator, Fig. 5 (a) is the anechoic chamber characteristic curve diagram of the midrange unit, and Fig. 5 (b) is the mid that processes the signal input to the midrange. FIG. 5 (c) is a characteristic curve diagram of a digital filter for a range, FIG. 5 (c) is a characteristic curve diagram of a mid-range anechoic chamber corrected by a digital filter, and FIG. 6 is a diagram of a sound quality adjusting apparatus for a speaker in a second embodiment of the present invention. FIG. 7 is a block diagram of a conventional hall sound system. 1 …… input terminal, 2 …… low pass filter, 3 …… A /
D converter, 4, 5, 6 ... Digital filter, 7, 8, 9 ... D
/ A converter, 10,11,12 …… Low pass filter, 13 …… Tweet terminal, 14 …… Mid range terminal, 15 …… Woofer terminal, 16 …… Interface, 17 …… Computer, 18 …… External storage device, 19 ... Display, 20 ...
… Light pen, 24 …… Speaker sound quality adjustment device, 36,37,
38 …… ROM.

Claims (1)

[Claims] 1. A first low-pass filter for reducing and filtering a music signal, an analog-digital converter for converting an output signal of the first low-pass filter into a digital signal, and a high-pass filtering for the converted digital signal. And a first digital filter for correcting the frequency characteristic of the high-pitched speaker unit, a second digital filter for band-filtering the digital signal and correcting the frequency characteristic of the middle-tone speaker unit, and a digital signal A third digital filter for low-pass filtering and correcting the frequency characteristics of the bass speaker unit, and a digital-analog converter for converting the outputs of the first, second, and third digital filters into analog signals, respectively. And the second, third, and second filters for low-pass filtering the converted analog signal, respectively. A low-pass filter, a calculation unit, and an interface unit, and the calculation unit realizes unit characteristic storage means for storing impulse responses of the high-pitched speaker unit, the mid-tone speaker unit, and the low-pitched speaker unit. Target characteristic input means for inputting a desired target amplitude frequency characteristic, Hilbert transforming means for Hilbert transforming the target amplitude frequency characteristic to calculate a phase characteristic, and amplitude and phase characteristics output from the Hilbert transforming means for time axis characteristics And an inverse Fourier transform means for converting into a time axis characteristic, and a tap coefficient calculation means for calculating a tap coefficient of each digital filter from the time axis characteristic and the impulse response output from the unit characteristic storage means, and the calculated tap The coefficient is transferred to the first, second, and third through the interface unit. A speaker sound quality adjustment device characterized by being transferred to a digital filter.