JPS61135298A - Sound quality adjusting device for speaker - Google Patents

Abstract

PURPOSE:To realize an object frequency characteristic with high accuracy by using a computer to control a tap factor given to a digital filter thereby making peak/dip flat on the sound pressure frequency characteristic of a speaker unit. CONSTITUTION:The computer 17 calculates a tap factor given to digital filters 4-6 and the tap factor is transferred to the filters 4-6 through an interface 16. A music signal inputted from a terminal 1 is subjected to low-pass filter by an LPF2, and inputted to the filters 4-6 via an A/D converter 3. The filters 4-6 execute band pass filtering in matching with the speaker unit, makes the steep peak and dip on the frequency characteristic flat and applies filtering to obtain an object frequency characteristic with high accuracy. The output of the filters 4-6 is outputted to terminals 13-15 via D/A converting sections 7-9 and LPFs 10-12.

Description

DETAILED DESCRIPTION OF THE INVENTION FIELD OF INDUSTRIAL APPLICATION The present invention relates to a speaker sound quality adjustment device for processing music signals input to a speaker system through a power amplifier in the field of hall acoustics.

BACKGROUND OF THE INVENTION In recent years, digital signal processing technology has made remarkable advances in the field of acoustics, particularly in the field of hall acoustics, where the development of digital mixers, digital equalizers, etc. has made it possible to create more detailed sounds. In addition, speaker systems use dedicated channel dividers to perform simple unit correction in addition to band division, and together with the digital signal processing technology mentioned above, sound quality adjustment technology in hall acoustic systems has made significant progress. It's coming.

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

FIG. 7 shows a system diagram of a hall acoustic system in a conventional embodiment. In FIG. 7, 21 is a microphone, and 22 is a mixer. 23 is an equalizer, and 39 is a channel divider. 26 is a driving power amplifier, 26 is a speaker system, 27,
28 and 29 are a tweeter for high frequency reproduction, a mid range for medium frequency reproduction, and a woofer for low frequency reproduction.

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

First, the music signal coming from microphone 21 is mixed into mixer 2.
2 and input to the equalizer 23. The equalizer 23 performs optimal filtering on the input music signal in consideration of the characteristics of the hall or viewing 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 divides the input music signal into bands and outputs it as a music signal for each unit. Additionally, each unit may have a filtering function for correcting the frequency characteristics or a delay function for phase matching. The output signals from the channel divider 39 are respectively driven by the power amplifier 26 and sent to the tweeter 2a, midrange 28 .
The signal is input to each speaker unit such as a woofer 29.

Problems to be Solved by the Invention However, in the above configuration, the filter incorporated in the channel divider 39 has a tone adjustment function composed of analog elements, and the equalizer 23 has a % octave adjustment function. Even if a speaker unit with a specific band width is used, it is difficult to change the sound pressure level of only one band width independently or suddenly, so the sound pressure frequency caused by each speaker unit may cause problems in terms of audibility. There was a problem in that it was not possible to completely flatten the sharp peaks and dips in the characteristics. Furthermore, even if a highly accurate digital equalizer is used as the equalizer 23, there is a problem in that it takes time to set up the equalizer.

In view of the above-mentioned problems, the present invention flattens the sharp peaks and dips in the sound pressure frequency characteristics of each speaker unit used in a speaker system, and also makes it possible to easily and quickly adjust the sound system at concert venues such as halls. To provide an excellent sound quality adjustment device for speakers that allows setting.

Means for Solving the Problems In order to solve the above problems, the speaker sound quality adjustment device of the present invention includes a first low-pass filter for low-pass filtering a music signal inputted from an input terminal, and a first low-pass filter for low-pass filtering a music signal inputted from an input terminal. and an analog-to-digital converter that converts the filtered music signal into a digital signal.

a first digital filter for high-pass filtering the digital signal and correcting the frequency characteristics of the high-pitched speaker unit; and a first digital filter for bandpass-filtering the digital signal and correcting the frequency characteristic of the mid-range speaker unit. a third digital filter for low-pass filtering the digital signal and correcting the frequency characteristics of the bass speaker unit; Second. a digital-to-analog converter for converting the outputs of the third digital filter into analog signals, and a second digital-to-analog converter for low-pass filtering the respective analog signals. and third and fourth low-pass filters, and the first and fourth low-pass filters. @21 The third digital filter is connected via an interface to the computer main body to which an external storage device and input device are connected.

According to the above-described configuration, the present invention provides information to each digital filter by a computer based on the frequency characteristic data of each speaker unit stored in the external storage device and the target frequency characteristic to be achieved inputted from the input device. Calculate the power tap coefficient. Tap coefficients are transferred from the computer to each digital filter through an interface. The music signal input through the input terminal is low-pass filtered by a low-pass filter, then converted into a digital signal by an analog-to-digital converter, and then input to each digital filter.

Each digital filter performs bandpass filtering suitable for each speaker unit, flattens sharp peaks and dips in the frequency characteristics of each speaker unit, and performs filtering to accurately achieve target frequency characteristics.

The output from each digital filter is converted into an analog signal by a digital-to-analog converter, and then low-pass filtered by a low-pass filter and output. Through the above-mentioned actions, it is possible to completely eliminate sudden peak dips in the sound pressure frequency characteristics caused by each speaker unit, which are audible problems, with simple and short-time operations, and to achieve the target frequency characteristics with high accuracy. becomes.

Embodiment Hereinafter, a speaker sound quality adjustment device according to an embodiment of the present invention will be described with reference to the drawings. FIG. 1 shows a block diagram of a speaker sound quality adjustment device 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 LP
3 is an analog-to-digital converter (hereinafter referred to as A/
4 and 5.6 are digital filters, and 7° and 8.9 are digital-to-analog conversion sections (hereinafter referred to as D/A converters).
13 is a tweeter terminal, 14 is a mid range terminal, 16 is a woofer terminal, and 16 is an interface for connecting to the computer 1.

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

First, FIG. 2 shows a system diagram of a hall acoustic system including the present invention, in which music signals input from a microphone 21 are input to a mixer 22. The signal is input to a speaker sound quality adjustment device 24 through an equalizer 23 for correcting an indoor sound field in a hall or the like. In Figure 1, the input music signal is L
After being low-pass filtered by the PF 2, the signal is converted into a digital signal by the A/D converter 3. The digital signal is input in parallel to a plurality of digital filters 4.5.6, and the tweeter 27° midrange 28.6 of each speaker unit shown in FIG. In addition to being divided into the frequency bands covered by the woofer 29, the sound pressure is filtered to flatten the peaks and dips in the sound pressure frequency characteristics of each speaker unit, and to obtain the frequency characteristics desired by the operator. like this.

Each digital filter 4. 5. The tap coefficient to be given to the interface 16 is calculated in advance by the computer and then
through each digital filter 4, 5. 8 memory, the algorithm will be described later. Digital filter 4. 5. The digital signal outputted from 6 is demodulated by D/A converter 7.8.9, then low-pass filtered by low-pass filter 10.11.12, and outputted from each output terminal 13-16.

In FIG. 2, the music signal output from the speaker sound quality adjustment device is amplified by a power amplifier 26 and drives each speaker unit constituting the speaker system 26.

Next, each digital filter 4, 6 . The procedure and algorithm for calculating the tap coefficients transferred to θ will be explained.

FIG. 3 is a diagram showing a digital filter tap coefficient calculation algorithm. First, in FIG. 2, a light pen 20 is used as a waveform input device. A target frequency characteristic is input using the display 19 (step 30). The input target frequency characteristics are amplitude information or real part information, which is subjected to Hilbert transform to obtain transmission characteristics, which are then divided into the bands that each speaker unit is responsible for (
Step 31). Furthermore, an inverse Fourier transform is applied to each band to obtain the impulse response of the target characteristic for each band (step 32). Next, in FIG. 2, all the impulse responses of each speaker unit recorded in the external storage device 18 are read (step 33), and convolution integration with the target characteristics divided into each corresponding band is performed (step 33). 34) Calculate tap coefficients and send them from the computer 17 to the interface 16 to each digital filter 4. 5. 6 memory (step 36).

As a specific example, Figure 4 shows how to calculate the mid-range digital filter tap coefficient 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 speaker system target frequency characteristic, which is input from the display 19 using the light ben 2o in FIG. $4
In the figure, this target frequency characteristic is set to a preset cutoff frequency F1. Characteristics divided into bands based on F2 and cutoff characteristics are also shown. Figure 6 (,) shows the anechoic chamber characteristics of the mid range unit.
The external storage device 18 shown in the figure is recorded as an impulse response including phase information. tlcs diagram (b) represents the tap coefficients transferred to the mid-range digital filter 6 on the frequency axis, and shows the mid-range target characteristics shown in FIG. It is obtained by convolution integral of the impulse response of the mid-range unit anechoic chamber characteristics shown in the figure (-). FIG. 6(C) shows mid-range anechoic chamber characteristics corrected by the digital filter 6 using the tap coefficients obtained in this manner. In FIGS. 4 and 6, Fl and F2 are cut-off frequencies, and FS' is a sampling frequency (Nyquist).

As described above, according to this embodiment, the LPF 2
The low-pass filtered music signal is converted into a digital signal by the A/D converter 3, and the digital signal is input to a plurality of digital filters 4 to 6 arranged in parallel, and each digital filter 4 to e is Through the interface 16, external storage device 18 and light pen 20. By connecting to a computer 17 connected to an input device such as a display and controlling the filter characteristics of each digital filter 4 to 6, a simple operation of inputting a target frequency characteristic can eliminate problems with hearing sensation. It is possible to flatten the peaks and dips in the sound pressure frequency characteristics caused by each speaker unit, and to quickly create sound that is suitable for the indoor environment.

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

FIG. 6 shows a speaker filter showing a second embodiment of the present invention, 7. 8. 9 is a D/A converter, 13 is a tweeter terminal, 14 is a midrange terminal, and 16 is a woofer terminal, which is the same as the structure shown in FIG. 1. The difference from the configuration shown in Figure 1 is that instead of connecting each digital filter to the computer through an interface, the RO
This is the point where M36, 37°38 are connected.

In the speaker sound quality adjustment device configured as described above, the tap coefficients of each digital filter transferred from the computer through the interface are stored in the ROM 36, 3.
7.38 is stored in memory. Therefore, in cases where the speaker system to be used in the field of equipment acoustics has been decided and will not be changed, the tap coefficients once calculated can be stored in ROM and accessed from this ROM. In addition to reducing the number of devices, the middle frequency range of each audio device is further divided into two or more bands, each of which has a digital filter 6, a D/A converter 8, and a low-pass filter 11.
It is also possible to use a system with four or more units.

Effects of the Invention As described above, the present invention applies filtering to the digital signal converted by the D/A converter to correct the frequency characteristics of each speaker unit of the speaker system and to achieve the target characteristics set by the operator. By providing a digital filter for controlling the sound pressure frequency characteristics of each speaker unit and controlling the tap coefficients given to the digital filter by a computer connected to the digital filter through an interface, peaks in the sound pressure frequency characteristics of each speaker unit that may be a problem in hearing sense can be eliminated.

It is possible to flatten the dip and achieve the target characteristics input by the operator in a short time with simple operations.

[Brief explanation of the drawing]

FIG. 1 is a block diagram of a speaker sound quality adjustment device according to the first embodiment of the present invention, FIG. 82 is a system diagram of a hall acoustic system using the present invention, and FIG. 3 is a digital filter tap coefficient calculation algorithm. Figure 4 is the speaker system target frequency characteristic curve input by the operator, Figure 6 (,) is the anechoic chamber characteristic curve of the midrange unit, and Figure 6 (g)) is the input to the midrange. FIG. 6(C) is a characteristic curve diagram of a digital filter for a mid-range range that processes signals; FIG. The block diagram of the adjustment device, FIG. 7, is a conventional hall acoustic system system diagram. 1...Input terminal, 2...Low pass filter, 3...A/D converter, 4. 5. 6
...Digital filter, 7. a, 9-・
-D/A converter, 10, 11゜12...Low pass filter, 13...Tweeter terminal, 1
4...Terminal for midrange range, 16...
Woofer terminal, 16...interface,
17... Computer, 18... External storage device, 19... Display, 20...
・-・Light pen, 24...Speaker sound quality adjustment device, 36. 37. 38...ROM0 Name of agent Patent attorney Toshio Nakao and 1 other person Ward/Castle Figure 3 Figure 4 jI Destruction Figure 5 Shoulder L collection

Claims (2)

[Claims] (1) A first low-pass filter that low-pass filters a music signal, an analog-to-digital converter that converts the output signal of the first low-pass filter into a digital signal, and a high-frequency filter that high-pass filters the converted digital signal and converts the output signal into a digital signal. a first digital filter for correcting the frequency characteristics of the midrange speaker unit; a second digital filter for bandpass filtering the digital signal and correcting the frequency characteristics of the midrange speaker unit; a third digital filter for performing area filtering and correcting the frequency characteristics of the bass speaker unit; and a digital-to-analog converter for converting the outputs of the first, second, and third digital filters into analog signals, respectively; A sound quality adjustment device for a speaker, comprising second, third, and fourth low-pass filters for respectively low-pass filtering converted analog signals. (2) The first, second, and third digital filters are connected to a computer using an interface, and the frequency characteristic data of the speaker unit recorded in an external storage device connected to the computer and the frequency characteristic data of the speaker unit connected to the computer are connected. The first, second, third
2. The speaker sound quality adjustment device according to claim 1, wherein the frequency characteristic of the digital filter is changed.