KR940002436B1 - Control apparatus and method of a digital tone - Google Patents

Abstract

The digital sound field controller and method therefor includes an ADC, a RAM for storing input signal, a DSP for separating input signal in four states to provide a presence, a display, a keyboard, a microcomputer for allowing the DSP to variably provide the presence to the input signal and transmitting display data, and an FDAC and an RDAC for restoring the four states of the DSP into analog signal, thereby varying the presence of signal.

Description

Digital sound field control device and method

1 is a block diagram according to the present invention.

2 is a functional block diagram of a DSP according to the present invention.

3 is a functional block diagram of the first and second filters.

4 is a functional block diagram of a first reflection sound generating unit.

5 is a block diagram according to the present invention of the keyboard.

6 is a conceptual diagram of the actual playing space.

7 is a reflection sound characteristic of the actual playing space.

8 is a reflection sound output characteristic of the reflection sound generating unit.

9 is a position view of a speaker and a listener in a sound field reproduction space according to the present invention.

10 is a characteristic diagram of change of direct sound, first reflection sound, and other reflected sound according to a delay value.

11 is a characteristic diagram showing the change in spacing between reflections of different hole sizes.

12 is a characteristic diagram of level change of reflected sound according to intensity.

13 is a flowchart for reading a key according to the present invention.

14 is a flowchart of parameter reading in accordance with the present invention.

15 is a flowchart for setting a program according to the present invention.

16 is an input level adjustment flowchart according to the present invention.

17 is a flowchart of output level adjustment according to the present invention.

18 is a flowchart of setting a hole size according to the present invention.

19 is a flowchart of distance setting between a sound source and a wall surface according to the present invention.

20 is a ring control flowchart according to the present invention.

21 is a flowchart of setting wall attenuation characteristics according to the present invention.

Figure 22 is a memory diagram in accordance with the present invention.

* Explanation of symbols for main parts of the drawings

101: ADC (Analog to Digital Convortor) 102: RAM

103: keyboard 104: micom

105: display unit

106: DSP (Digital Signal Processor) 107: FDAC (Front DAC)

108: RDAC (Rear DAC) 200: matrix circuit

DR: Delay

The present invention relates to a sound field control apparatus and method, and more particularly, to a digital sound field control apparatus and method for converting an analog signal into a digital scene to impart a sound field.

In general, the sound field control device is realized by a circuit using an element such as an analog delay element, and because it reproduces only the primary simple sound, the sense of presence is considerably lacking.

Accordingly, it is an object of the present invention to provide a sound field control device that enables the sound field control function to be excellent in its characteristics by using digital signal processing and to generate a high order reflected sound, and to allow the user to arbitrarily change the characteristic.

Another object of the present invention is to provide a method for generating a higher order reflected sound when the sound field is controlled and allowing the user to arbitrarily adjust the sound field change state.

1 is a block diagram according to an embodiment of the present invention, the ADC 101 for receiving a stereo audio signal and converting it into a digital signal, a RAM 102 for storing and reproducing an input signal for a predetermined time, and the ADC 101. DSP which receives an output and exchanges data with the RAM 102, receives predetermined coefficient data, thereby separating input signals into four states, that is, FR ', FL', RR ', and RL' and controlling them to give a sound field. 106, a display unit 105 for receiving and displaying predetermined display data, various function keys for a sound field control function, and a keyboard 103 for generating and outputting corresponding digital signals when they are selected, and the keyboard ( After receiving the key data from 103 and processing the corresponding data, the coefficient data is output to the DSP 106 so that the DSP 106 can vary the sound field to the input signal according to the key data. The microcomputer 104 transmits display data to the display unit 105 so that the display unit 105 can display the state of the key data, and the FDAC which receives the four-state output of the DSP 106 and outputs the analog signal. 107) and the RDAC 108.

5 is a key configuration diagram of the keyboard 103 according to the present invention, and includes a delay time key 501, a hole size key 502, an input level key 503, a low pass filter key 504, and a high pass filter key ( 505, an output level key 506, a program key 507, an intensity key 508, a parameter up key 509, and a parameter down key 510.

13 to 22 are flow charts according to the present invention, in which a key reading process of outputting initialization data to each part at power supply and then reading the key when there is a key input and branching accordingly, and in the key reading process A parameter determination process for determining a parameter at the time of the parameter up key or a parameter down key as a result of reading a key input and branching accordingly; and reading a program key as a result of reading the key input in the key reading process, The program process of setting the sound field state corresponding to each place of the actual playing space according to the key input when it is determined by the program, and the result of reading the key input in the key reading process, is read by the input level key or the parameter is read in the parameter reading process. An input level control process of controlling an input signal level when it is determined as an input level; An output level control process for controlling the output level if the input key data is determined as an output level key in the reading process or a parameter is determined as an output level in the parameter reading process, and the input key data is determined as a hole size key in the key reading process, or In the parameter determination process, if the parameter is determined as the hole size, the hole size control process for controlling the hole size acoustically by controlling the time interval of each reflection sound, and in the key reading process, the input key data is determined as the delay time key or the parameter is determined. If the parameter is determined as the delay time in the process of controlling the distance between the direct sound and the reflection sound wall distance setting process for controlling the distance between the sound source and the wall, and in the key reading process the input key data is determined as the intensity key or Parameter is displayed as intensity during parameter determination A sound control process that controls the pattern of each reflection sound so as to acoustically control the sound of the reflection sound, and in the key reading process, the input key data is determined as the low pass or high pass filter key, or the parameter is read low or high pass in the parameter reading process. If it is determined to be, the wall attenuation control process is performed to control the attenuation characteristics of the low or high range by acoustically controlling the attenuation characteristics of the low or high range by the material of the reflective surface of the actual performance space.

FIG. 23 is a memory diagram according to the present invention, in which memory regions within the microcomputer 104 are set, and each region is allocated to perform the present invention.

Before describing an embodiment of the present invention, a brief review of the principle aspects of the present invention is given.

In a performance space such as a real hall, the listener has the direct sound (DR) generated from the sound source (S) as shown in FIG. 6, as well as the reflected sound ER0, ER1, ER2, ER3, ..., etc. reflected from the wall, floor, or ceiling of the hall. Will be reached. Therefore, the listener hears the reflection of the performance space in addition to the direct sound, which acts as a factor that makes the listener feel a sense of presence. In addition, in the actual playing space, the reflection sounds ER0, ER1, ER2, ER3, ... are different for each playing space, and the listener can feel the difference in each playing space.

7 shows the impulse response of the performance space, where ER1, ER2, ER3, ... appear after time t1 after the direct sound DR. If the reproducing apparatus can produce the same pattern of reflection sound as the actual playing space, that is, the reflection pattern as shown in FIG. 7, the sound will be the same as the sound of the actual playing space. The present invention applies this principle.

Hereinafter, embodiments of the present invention will be described in detail with reference to FIGS. 1 to 22.

In Fig. 1, the analog input signals L and R are input to the ADC 101 and converted into digital audio data L 'and R', and the output thereof is input to the DSP 106. In addition, the DSP 106 receives each coefficient data from the microcomputer 104 controlled by the keyboard 103, thereby giving a sound field to the digital audio data L ', R' of the ADC 101 and the state thereof. To control. In order for the DSP 106 to perform a sound field control function, the DSP 106 must have a function of delaying input data by a predetermined time. The RAM 102 exchanges data with the DSP 106 and inputs the data to the DSP 106. This function saves the input signal for a certain time and outputs it again to give a sound field to the signal. The keyboard 103 additionally has the keys shown in FIG. 5 as well as the keys typically required for control of the DSP 106, and when they are selected, outputs corresponding digital data to the microcomputer 104. FIG. do. The microcomputer 104 controls the DSP 106 and simultaneously outputs various parameters to the display 105 so that the display 105 can display them.

On the other hand, the output of the DSP 106 is divided into the front channels FL ', FR' and the rear channels RL ', RR', and the front channel output digital data is input to the FDAC 107 to output analog audio signals. The back channel output digital data is input to the RDAC 108 and restored to an analog audio signal.

The operation and function of the DSP 106 are shown in the circuit diagrams of FIGS.

The DSP 106 is one digital signal processor chip that is executed under the control of the microcomputer 104, but a person having ordinary knowledge in the art may fully implement the device other than the DSP 106. That is, referring to FIGS. 2 to 4, the circuit can be sufficiently replaced with a discrete circuit or an ASIC device. The DSP 106 actually receives coefficient data by the microcomputer 104 having the flow of the thirteenth through twenty-two degrees, and each function is set accordingly.

Therefore, the function of the DSP 106 will be described with reference to FIGS.

The input digital audio signals L 'and R' are input to the multipliers m1 and m2 and their levels are controlled. At this time, the overflow due to over input is prevented. The outputs of the multipliers m1 and m2 are input to the matrix circuit 200 consisting of multipliers m3 to m6 and adders AD1 and AD2, and the matrix circuit 200 is configured to adjust the left and right level signals of the signal to be subjected to the reflection sound signal processing. Generate sum and difference signals.

When the control coefficient value of the multipliers m3 to m6 is m3 = m4 = m5 = m6 = 0.5, the L channel output of the matrix circuit 200 becomes L "= 0.5L '+ 0.5R', and the R channel output It becomes R "= 0.5L '+ 0.5R', and it becomes the sound which L and R signals added at the same ratio.

When the control coefficients of the multipliers m3 to m6 are set to m3 = m6 = 1 and m4 = m5 = -1, L "L'-R'and R "R'-L' become L and R signals. Is the difference beep. When the control coefficient values m3 = m6 = 1 and m4 = m5 = 0, L "= L 'and R" = R' are output, and L and R signals are output as they are. In this way, the matrix circuit 200 simulates mixing of the sound in the actual playing space by appropriately mixing the coefficient value states of the multipliers m3 to m6 and the signals L and R of the source sound generating the reflection sound according to the adders AD1 and AD2. Has the function. The output L ", R" of the matrix circuit 200 controlled in this way is input to the first and second filters 210 and 220 for controlling the frequency output of the source sound, and the first and second filters 210 and 220 are As shown in FIG. 3, the characteristic is determined as a high or low filter according to the coefficient data A10 to B21 supplied from the microcomputer 104 having two cascaded IIR digital filters having the same configuration. do. When the sampling frequency is fs and the cutoff frequency is fc, π = 3.141592, and Tx = TAN ( π × fc / fs), when the first and second filters 210 and 220 have a function of a high pass filter, The coefficients A10, A11 and B11 are respectively,

A10 = 1 / (1 + Tx)... … … … … … … … … … … … … … … … … … … One

A11 = -1 / (1 + Tx)... … … … … … … … … … … … … … … … … … 2

B11 = -1 (Tx-1) / (1 + Tx)... … … … … … … … … … … … … … 3

It is decided.

Further, when the first and second filters have the function of a low filter, the coefficients A20, A21, and B21 of the LPF are respectively,

A20 = Tx / (1 + Tx)... … … … … … … … … … … … … … … … … … … 4

B21 = -Tx / (1 + Tx)... … … … … … … … … … … … … … … … … … 5

B21 =-(Tx-1) / (Tx + 1)... … … … … … … … … … … … … … … … 6

It is decided.

The first and second filters 210 and 220 simulate the wall characteristics of the actual sound reproduction space by controlling the frequency characteristics of the source sound, and the wall materials of the actual performance space absorb the low sound or the high sound. By properly controlling the characteristics of the first and second filters 210 and 220 with a low pass filter or a high pass filter, the characteristics of the wall surface of the playing space can be simulated. In FIG. 2, the output of the first filter 210 is output to the first and third reflection sound generators 230 and 250, and the output of the second filter 220 is output to the second and fourth reflection sound generators 240 and 260. Is entered. The first to fourth reflection sound generating units 230 to 260 have the same configuration, but each coefficient is different. The reflection sound generating units 230 to 260 are controlled by the microcomputer 104 of FIG. 1 and operate in conjunction with the RAM 102 to perform functions similar to those of FIG. 4.

Since the reflection sound generating units 230 to 260 all have the same configuration, the function will be briefly described with reference to FIG. 4 as the first reflection sound generating unit 230.

The delay DL consists of the RAM 102 controlled by the DSP 106.

The signal data input at the time t (W10) designated as the data write address W10 of the retarder DL is output at the time t (R10) designated as the data lead time address R (10), and this output signal is reflected sound. It is added by the adder AD7 via the multiplier G10 for level control. The signal at this time is a signal obtained by multiplying the multiplication coefficient G10 by a signal delayed by time tΔ10 = t (R10) -t (W10), and subsequently being designated by the data lead address R11 of the delay unit DL. The signal output to t (R11) is added by the adder AD7 via multiplier G11. The signal at this time is a signal obtained by multiplying the signal delayed by the time tΔ11 = t (R11) -t (W10) by the multiplication factor G11. In the same principle, the signals output at the times t (12) to t (21) designated by R11 to R21 are calculated by G12 to G21 and added to the adder AD7. As a result, the output of the first reflection sound generator 230,

It can be represented as

The function of the reflection sound generators 230 to 260 can be easily understood by comparing FIG. 7 and FIG. 8, and the outputs of the reflection sound generators 230 to 260 are the same as those of FIG. 8. It is the same as after the first reflection sound having gain ER0 at time t0 in the playing space of FIG.

Therefore, when an audio signal is reproduced in a predetermined unit having the characteristics of FIG. 8, a sense of presence similar to that in the actual performance space shown in FIG. 6 can be reproduced.

The outputs FR ″, RL ″, and RR ″ of the second to fourth reflection sound generators 240 to 260 may be defined as follows.

In the present invention, the processing of the reflection sound generation divided into four channels of the front FR, FL and the rear RR, RL is performed in order to simulate countless units because the listener 603 hears the reflection sound from front, rear, left, and right directions in FIG. In order to obtain characteristics similar to those of FIG. 6, the sound field is synthesized using four speakers as shown in FIG. That is, in FIG. 9, the FL speaker 901, the FR speaker 902, and the RL speaker 904 and the RR speaker 905 divide the reflection sounds reproduced in the respective channels, so that when these four reflection sounds are combined, the listener ( At position 603, a sound field similar to the actual playing space is reproduced. The outputs of the first to fourth reflection sound generators 230 to 260 are respectively supplied to the multipliers m9, m10, m14, and m15 for reflecting sound level control for simultaneously controlling the level of the entire reflected sound, and the outputs of the multipliers m9 and m10 are directly The outputs of the sound level control multipliers m8 and m11 are added to the adders AD3 and AD4. This is because the listener 603 has a direct sound DR coming directly, in which case the subject of the sound is the direct sound DR and the reflected sound serves as an auxiliary signal for giving a sound field. Multipliers m12, m13, m16, and m17 are for controlling the output level, and the degree of attenuation is determined according to the coefficient data transmitted from the microcomputer 104.

Each multiplier and each filter coefficient of the first to fourth degrees are supplied by the microcomputer 104 as mentioned above, and the microcomputer 104 stores them in an internal ROM that stores the contents of the 13 to 21 degrees. It accesses and supplies to each said part.

The state in which the coefficient data is stored in the internal ROM of the microcomputer 104 is shown in FIG. 22, and a description thereof is considered by those skilled in the art to understand how the microcomputer uses. Omit the description.

Hereinafter, the operation of the microcomputer 104 will be described with reference to the flows of FIGS. 13 to 21.

FIG. 13 is a flowchart of a key reading process for recognizing when a user inputs a predetermined key. When the power of the system is supplied, the microcomputer 104 firstly initializes the data (default mode) to the DSP 106 in step 13a. Data) and waits for key input from the keyboard 103 in step 13B. In this case, when the parameter up or down key is input, the microcomputer 104 analyzes the input key while performing steps 13c to 13k and branches to the corresponding process.

14 is a parameter reading process performed when the input key data is determined as a parameter up or parameter down key in the key reading process. The microcomputer 104 performs steps 14b to 14i after checking the current parameter in step 14a. According to the result of the investigation in step 14a, the process branches to the corresponding parameter. If no key is input, the microcomputer 104 returns to the key input process and continues checking key input. The parameter corresponds one-to-one to a key according to the present invention of the keyboard 103.

FIG. 15 is a flowchart of a functional program process for setting an initial sound field state of the DSP 106 when a key input is determined as a program key in the key reading process or a parameter is determined as a program key in the parameter determining process. A mode, that is, sound field data capable of reproducing an arbitrarily assumed sound field space such as a concert hall or a church, is supplied to the DSP 106 by using the parameter up or parameter down keys, and the digital signal to which the DSP 106 is input. According to the sound field corresponding to the assumed sound field space can be given. The sound field data is data to be supplied to each of coefficients W10 to W40, m8 to m11, R10 to R57, m3 to m6, G10 to G57, and m14 to m15 of the equivalent circuits of the DSP 106. Refers to data having the power of FIG. 22 stored in the internal ROM of the microcomputer 104.

First, the microcomputer 104 checks whether a parameter up or a parameter down key is input in step 15a. If it is not input, the micom 104 returns to the key input process, and if there is an input of the corresponding key, whether the parameter up key or a parameter down key is input in step 15b. Check it. In the case of the parameter up key, the microcomputer 104 increases the predetermined variable PRG value corresponding to the number by one step in step 15c and checks whether the value is greater than or equal to 10 and is greater than or equal to (15d). Step), in step 15e, set PRG = 10, otherwise keep the PRG value as it is. In addition, in step 15b, if the input key is determined to be a parameter down key, the microcomputer 104 proceeds to step 15f to decrease the PRG value by one step, and then checks whether the value is less than or equal to zero. On the other hand, if PRG = 1 is set in step 15h and not greater than or equal to 0, the PRG value is kept as it is.

After performing step 15e or step 15h, the microcomputer 104 proceeds to step 15i, and each data corresponding to the PRG value in its internal ROM is W10 to W40, m80 to m11, and R10 to R57. , m3 to m6, G10 to G57, and m14 to m15 are read out and supplied as coefficient data to each of the second to fourth degrees in step 15j. When the transmission is completed, the microcomputer 104 checks whether another random key is input in step 15k. When the predetermined key is input, the microcomputer 104 returns to step 15a to repeat the flow. However, if no key is input, the microcomputer 104 returns to the key reading process and waits for key input.

FIG. 16 is a flowchart of an input level adjusting process for controlling an input level of the DSP 106 when an input key is determined as an input level key in the key reading process or a parameter is determined as an input level in the parameter reading process. In order to prevent an overflow from occurring due to an over-input to 106, first, the microcomputer 104 checks whether a parameter up or down key is input in step 16a. If any one of them is input, the microcomputer 104 determines whether it is a parameter up key or a parameter down key in step 16b.

At this time, if the parameter up key, the microcomputer 104 increases the predetermined variable INLVL value corresponding to the input level of the DSP 106 in step 16c by one step and checks whether the value is greater than or equal to 30 in step 16d. If the value is greater than or equal to, the INLVL value is set to 30 in step 16e, and if the value is smaller than or equal to, the value is kept as it is.

In addition, if the input key is determined as the parameter down key in step 16b, the microcomputer 104 decreases the variable INLVL value by one step in step 16f and checks whether the value is less than or equal to 0 in step 16g. If it is less than or equal to, the variable INLVL value is set to 0 in step 16h, and if it is not less than or equal to, the INLVL value is maintained as it is.

After performing steps 16e and 16h, the microcomputer 104 reads data m1 and m2 corresponding to the INLVL value from its internal ROM and transmits the data m1 and m2 to the DSP 106 to transmit the DSP ( Set the input level of 106).

Thereafter, the microcomputer 104 checks whether another predetermined key is input, and if no key is input, returns to the key reading process, waits for input of a new key, and returns to step 16a when a predetermined key is input. The above operation is repeated.

FIG. 17 is a flowchart of an output level adjusting process for controlling an output level of the DSP 106 when an input key is determined as an output level key in the key reading process or a parameter is determined as an output level in the parameter reading process. This function allows you to arbitrarily adjust the level of the effect sound according to your taste.

First, the microcomputer 104 checks whether a parameter up or a parameter down key is input in step 16a. When those keys are input, it checks whether a parameter up key is input in step 16b.

At this time, when the parameter up key is input, the microcomputer 104 increases the predetermined variable OUTLVL value corresponding to the output level of the DSP 106 by one step in step 17c and checks whether the value is greater than or equal to 30 in step 17d. .

At this time, if the OUTLVL value is greater than or equal to 30, the microcomputer 104 sets the variable OUTLVL value to 30 in step 17e, otherwise, the value remains as it is.

On the other hand, if the input key is determined to be a parameter down key in step 17b, the microcomputer 104 decreases the variable OUTLVL value by one step and checks whether the value is equal to 0 in step 17g. At this time, if the variable OUTLVL value is equal to or smaller than 0, the microcomputer 104 sets the variable OUTLVL value to 0 in step 17h, and otherwise maintains the value.

After performing step 17e or step 17h, the microcomputer 104 reads each of m12, m13, m16, and m17 corresponding to the variable OUTLVL value from an internal ROM and outputs the same to the DSP 106. Thereafter, the microcomputer 104 checks whether there is another key input, and if no key is input, returns to the key drawing process. If any key is input, the microcomputer 104 repeats the above-described function according to the key input in step 17a. Done.

FIG. 18 controls the time interval between each reflection sound when the DSP 106 gives a sound field to the input signal when the input key data is read with a hole size key in the key reading process or the parameter is read with a hole size in the parameter reading process. As a flowchart of a hole size setting process for acoustically controlling the size of a hole, the microcomputer 104 first determines whether a parameter up key or a parameter down key is input in step 18a. It returns and waits for a key input standby state, and if any one of them is input, it checks whether it is a parameter up key in step 18b. In this case, if the parameter key is correct, the predetermined variable HSIZE value corresponding to the hole size is increased by one step in step 18c, and then, in step 18d, it is checked whether the value is greater than or equal to 20. If the value is greater than or equal to 20, the microcomputer 104 sets the variable HSIZE value to 20 in step 28e and maintains the HSIZE value as it is not less than or equal to 20.

In addition, in step 18b, if it is not the parameter up key, in step 18f, the variable HSIZE value is decreased by one step, and it is checked whether the value is less than or equal to five. At this time, if the variable HSIZE value is less than or equal to 5, the microcomputer 104 sets the HSIZE value to 5 in step 18h. If the variable HSIZE value is less than or equal to 5, the microcomputer 104 maintains the HSIZE value as it is.

In step 18i, after performing step 18e or step 18h, the microcomputer 104 performs data read time address value data of the HHS value and the data corresponding to the HSIZE value in the internal ROM in step 18i. R (10) to R (57) are read out.

When the HS value is read, the microcomputer 104 performs a predetermined operation with the data read time address values R10 to R57 of the first to fourth reflection sound generating units 230 to 260. In step 18j, the variable n for counting the data read time address value of the reflection sound generators 230 to 260 is set to 10, and the variable s for separately counting the first to fourth reflection sound generators 230 to 260. Set to 0. In step 18k, the predetermined variable RX (n) is defined as RX (n) = R (n), and in step 18l, the time change Δt is defined as the predetermined variable TR (n) and its value is RT (n) = R ( Calculate as n + 1) -R (n).

In step 18m, RX (n + 1) is calculated as RX = RT (n) × HS + RX (n), and the result of RX (n + 1) becomes the data read time address value of the newly generated reflection sound generating unit. . After this calculation, the microcomputer 104 increases n by one step in step 18n and checks whether n = 21 + s in step 18o. At this time, if n = 21 + s, the microcomputer 104 returns to step 18l to repeat the above calculation. This is for calculating a data read time address value for each of the first to fourth reflection sound generating units 230 to 260. For example, when a new data read address value for the first reflection sound generator 230 is calculated, R10 to R21 and HS = 2, since first n = 10, RX (10) = R (10).

TR (11) = (R (11) -R (10) and RX (11) = RT (10) x HS + RX (10) = [R (11) -R (10)] x 2 + RX (10).

If n is increased by one step, n = 11, and n = 21 + s = 21, so if RT (11) = RT (12) -R (11) calculation is performed again, RX (12) = RT (11) × HS + RX (11) = [R (12) -R (11)] × 2 + RX (11).

In the same manner, when n = 20, RX (21) calculation is performed, and since n = 21, s = s + 12 = 33. As a result, the calculation of the R10 to R21 (21) of the first reflection sound generator 230 is completed, and the new data read time values RX (10) to RX (21) are calculated.

After calculating the new data lead time address of one reflection sound generation unit, that is, the first reflection sound generation unit 230 as in the above example, the microcomputer 104 returns s and n to s = 12, in step 18p. n = 22 is defined, and in step 18q, it is checked whether s is s = 48 and it is checked whether all calculations up to the fourth reflection sound generator 260 are completed. In this case, if s is not 48, the microcomputer 104 determines that the data read address values of all the reflection sound generating units 230 to 260 have not been calculated, and RX (n) is set to R (n) in step 18r. Replace with RX (22) = R (22) and return to step 18l. This time, the data read time address values RX (22) to RX (33) of the second reflection sound generating unit 240 are generated. Calculate

After calculating all data reading times of the first to fourth reflection sound generating units 230 to 260 in this manner, the microcomputer 104 branches from the eighteenth step to the eighteenth step, where R (n) = RX ( n), the values of RX (10) to RX (57) calculated above are changed to R (10) to R (57) and transmitted to the DSP 106 as coefficient data in step 18t.

FIG. 19 shows the distance between the sound source and the wall acoustically as controlling the time interval between the direct sound and the first reflection sound when the input key is determined as the delay time key in the key reading process or the parameter is determined as the delay time in the parameter reading process. Delay time setting flow chart for controlling.

The microcomputer 104 determines whether a parameter up or down key is input in step 19a as a first step and determines whether the parameter up key is in a next step 19b. At this time, if the parameter up key is correct, the microcomputer 104 increases the predetermined variable DLY corresponding to the delay value by one step in step 19c and checks whether the value is greater than or equal to 100 in step 19d. In this case, if the DLY value is greater than or equal to 100, the microcomputer 104 sets the variable DLY value to 100 in step 19e. Otherwise, the microcomputer 104 retains the value of step 19d.

On the other hand, if it is determined that the parameter up key in step 19b, the microcomputer 104 proceeds to step 19f to decrease the variable DLY value by one step and check whether the value is less than or equal to zero. At this time, if the value of the variable DLY is less than or equal to 0, the microcomputer 104 sets it to 1 in step 17h, otherwise, maintains the value in step 19f.

The operation from step 19b to step 19h is to increase or decrease the delay value between the reflection sounds corresponding to the DLY value according to the input of the parameter up or down key.

After performing step 19e or step 19h, the microcomputer 104 performs predetermined data RX corresponding to the OUTLVL value of FIG. 17 in the internal ROM and data read time data R of the reflection sound generating units 230 to 260. FIG. Read (10) -R (57).

Thereafter, the microcomputer 104 sets the predetermined variable n to n = 10 in step 19j, and performs steps 19k to 19m to the data read time data values R (10) to R (57). The data RX corresponding to the OUTLVL value is added. After this, the microcomputer 104 transmits the newly calculated values R (10) to R (57) to the DSP 106 as coefficient data. Then, if another predetermined key is input, the process returns to step 19a to repeat the above operation according to the key input, and if no key is input, returns to the key reading process and waits for the key input.

FIG. 20 illustrates that when the key input is determined as the intensity key in the key reading process or the parameter is determined as the intensity in the parameter reading process, when the DSP 106 gives a sense of presence to the input signal, the sound is controlled by controlling the size of each reflected sound pattern. It is a ring control process to control the ringing of reflected sound.

First, if a parameter up or parameter down key is input in step 20a of the microcomputer 104, it is checked whether it is a parameter up key in step 20b.

In this case, if the parameter is an up key, the microcomputer 104 proceeds to step 20c to increase the predetermined variable ITS corresponding to the intensity by one step and checks whether the value is greater than or equal to 20 in step 20d. And wherein if the variable ITS value is greater than or equal to 20, the micom 104 sets the variable ITS value to 20. However, if the ITS value is less than or equal to 20, the microcomputer 104 maintains the variable ITS value as the value of step 20c.

If the parameter down key is not input in step 20b and the parameter down key is input, the microcomputer 104 decreases the variable ITS value by one step in step 20f, and the value is less than 0 in step 20g. If it is equal or less than or equal to, it is set to 0 in step 20h, otherwise the value of step 20f is kept as it is.

After performing step 20e or step 20f, the microcomputer 104 performs a delay signal of the data CX and the reflection sound generator 230 to 260 corresponding to the variable ITS value in its internal ROM in step 20i. Read the multiplier coefficient values G10 to G57 for level control.

In step 20j, a predetermined variable n is set to n = 10, and steps 20k to 20m are performed to calculate the coefficient value G (n) corresponding to the variable as G (n) + CX to generate the reflection sound generator. CX is added to level control multiplier coefficient values G10 to G57 of 230 to 260, respectively.

After completing the addition, the microcomputer 104 transmits the newly calculated multiplier coefficient value to the DSP 106 in step 20n. Then, if any key is input, the microcomputer 104 returns to the first step 20a to repeat the above-described operation. However, if no key is input afterwards, the microcomputer 104 returns to the key reading process and remains in the key input standby state.

FIG. 21 is a low or low range when the DSP 106 imparts a sound field to an input digital signal when an input key is determined as a high pass or low pass filter key in the key reading process or a parameter is determined as a high pass or low pass filter in the parameter reading process. As a wall attenuation control process for adjusting the high attenuation characteristics to control the sound attenuation characteristics caused by the material of the reflective surface acoustically assumed to be a real performance space, first, the microcomputer 104 performs the parameter up or down in step 21a. The key is entered is checked, and then in step 21b it is checked whether he is a parameter upkey. In this case, if it is determined as a parameter up key, the microcomputer 104 increases the predetermined variable FLT value corresponding to the low pass or high pass filter by one step and checks whether the value is greater than or equal to 13 in step 21c.

At this time, if the FLT value is greater than or equal to 13, the microcomputer 104 sets the FLT value to 13 in step 21e, otherwise, maintains the FLT value at the value of step 21c.

In addition, if it is determined in step 21b that the parameter is not a key up key, the microcomputer 104 decreases the variable FLT value by one step in step 21f, checks whether the value is less than or equal to zero, and determines that the value is less than zero. If it is equal, the microcomputer 104 sets the FLT value to 0, and if it is not greater than or equal to 0, the microcomputer 104 maintains the FLT value at the value of step 21f.

After completing step 21e or step 21h, the microcomputer 104 checks whether the parameter is a high pass filter in step 21i. In this case, if the parameter is a high pass filter, the microcomputer 104 branches to step 21j to read HPF data corresponding to the FLT value from its internal ROM, and then, in step 21k, A10, A11, and FIG. Transmit with count data corresponding to B11. In this case, if the parameter is not a high pass filter but a low pass filter, the microcomputer 104 branches to step 21l, this time reading data corresponding to the low pass filter from the internal ROM to the DSP 106 in step 21n. It transmits the coefficient data corresponding to A20, A21, B21 of FIG.

After performing step 21k or step 21n, the microcomputer 104 checks whether any key is input again. If no key is input, the microcomputer 104 returns to the key input checking process and waits for key input. If a predetermined key is input, the process returns to step 21a to repeat the above operation.

While the operation and configuration of the present invention have been described above, it is to be understood that the present invention is not limited to the above description but may be changed without departing from the gist of the invention as necessary.

As described above, the present invention converts an input signal into a digital signal and then inputs it to the DSP 106 to control it through the microcomputer 104, thereby providing a sound field to the input signal. The use of the key 103 has the advantage that the sound field state can be arbitrarily changed.

In addition, since it is possible to generate a higher degree of reflection, there is an advantage that the sound field feeling is further advanced compared to the prior art, thereby providing a more vivid sound field feeling.

Claims (4)

In the audio sound field control device, an ADC (101) for converting a stereo audio signal into digital audio data, a RAM (102) for storing and reproducing an input signal for a predetermined time, and receiving the output of the ADC (101) A DSP 106 for receiving predetermined coefficient data while exchanging data with the RAM 102, thereby separating input signals into four states and controlling them to give a sound field; a display 105 for receiving and displaying predetermined display data; A keyboard 103 having various function keys for a sound field control function and generating and outputting corresponding digital signals when they are selected, and receiving the key data from the keyboard 103 and processing the corresponding key data accordingly, the DSP 106 Outputting the coefficient data to the DSP 106 allows the DSP 106 to give the input signal a variable sound field feeling according to the key data and sell the key data. The microcomputer 104 transmits the display data so that the dock state is displayed on the display 105, and the FDAC 107 and the RDAC 108 receive the four-state output of the DSP 106 and reduce the analog signal to an analog signal. Digital sound field control device characterized in that the configuration. A sound field control apparatus comprising: key reading means for outputting initialization data to each part at a time of high power, and then reading a key when there is a key input, and for branching accordingly; A parameter discriminating means for discriminating and branching according to a parameter corresponding to a parameter down key, and when a parameter determined by a program key in a key input reading of the key reading means or a parameter determined by the parameter discriminating means is programmed, each place of a real playing space Program means for setting a sound field state corresponding to the key input, and input for controlling the input signal level when the parameter determined by the input level key in the key input reading of the key reading means or the parameter determined by the parameter reading means is determined as the input level. A level control means and a key input read output level key of the key read means; Output level control means for controlling an output level if the parameter is determined to be an output level in the parameter reading means, and if a parameter is determined to be a hole size in the key input reading of the key reading means or reading from a pole size key; Hall size control means for acoustically controlling the hole size by controlling the time interval of the reflected sound; and if the parameter is determined as the delay time in the key input reading of the key reading means or the parameter is determined as the delay time, the direct sound and Reflected wall distance setting means for controlling the distance between the sound source and the wall acoustically by controlling the distance between the reflected sound, and if the parameter is determined as the intensity key in the key input reading of the key reading means or the parameter in the parameter determining means, By controlling the pattern size of the reflections, acoustic reflections are suppressed. And a low pass and a high pass in the ring control means and the key reading means of the key reading means, or when the parameter is determined as the low pass and high pass in the parameter reading means, the attenuation characteristics of the low pass and the high pass are controlled to perform acoustic performance. A digital sound field control device comprising wall attenuation control means for controlling the low and high frequency attenuation characteristics caused by the material of the reflective surface of the space. In the sound field control method, a key reading process of outputting initialization data to each part at power supply and then reading a key when there is a key input and branching accordingly, and a parameter up or down as a result of reading the key input in the key reading process. Parameter down key Determines the parameter at that time and branches accordingly, and reads the key input in the key reading process and reads the program key or the parameter in the parameter discriminating process. A program process for setting a sound field state corresponding to each location according to a key input, and an input signal level when a key input is read as a result of reading a key input in the key reading process or when a parameter is determined as an input level in the parameter reading process. An input level control process for controlling the input, and an input during the key reading process If the key data is determined as an output level key or the parameter is determined as an output level in the parameter reading process, an output level control process for controlling the output level, and in the key reading process, the input key data is determined as a hole size key or in the parameter discrimination process. When the parameter is determined to be the hole size, the hole size control process for controlling the hole size acoustically by controlling the time interval of each reflection sound, and in the key reading process, the input key data is determined as the delay time key or the parameter is determined in the parameter determination process. If the delay time is determined, the distance between the direct sound and the reflected sound is controlled so as to control the distance between the sound source and the wall acoustically. The distance between the sound source and the wall is set; and in the key reading process, the input key data is identified as the intensity key or the parameter discrimination process. If the parameter is determined by intensity If the input key data is determined as a low pass or high pass filter key in the key reading process or the parameter is determined as low pass or high pass in the key reading process A digital sound field control method comprising: a wall attenuation control process that controls the attenuation characteristics of a low range or a high range and acoustically controls the attenuation characteristics of a low range or a high range by a material of a reflective surface of a real performance space. The digital attenuation multiplier (m1, m2) for controlling the amplitude of the left and right digital audio data of the ADC (101), and the input attenuation multiplier (m1, m2). A matrix circuit unit 200 for logically calculating the sum and difference of the level signals required for the reflected sound signal processing by the output of the first and second matrix signals, and for simulating wall characteristics of the playing space. The first and second filters 210 and 220 for filtering the first and second matrix signals and the reflection coefficients of the first and second filters 210 and 220 are respectively included in the filtered first and second matrix signals of the first and second filters 210 and 220. The magnitudes of the first to fourth reflection sound generators 230 to 260 and the four to four reflection sound generators of the first to fourth reflection sound generators 230 to 260 which output four-way reflection sounds of the front left and right channels and the rear left and right channels, respectively. Comprising a logic operation unit that outputs four-way reflection sound in the actual playing space Digital sound field control device as claimed.

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