JPH11163652A - Control method for effect adjusting acoustic device and recording medium recording its control program - Google Patents

Abstract

PROBLEM TO BE SOLVED: To prevent noises from being mixed with output sound caused by abrupt level changes due to effect adjustment. SOLUTION: This method carries out a digital signal processor transmission preparation step S22 which sets data that derives coefficient data from a microcomputer inside based on a level that has undergone effect adjustment, a coefficient data deciding step which decides coefficient data based on the level that has undergone effect adjustment, a sending step which gradually sends the decided coefficient data to an effect processing means 9 and an effect adjustment finishing step which finishes the effect adjustment in an input processing step S21.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention records a control method of an effect-adjusting sound apparatus for preventing noise from being mixed when an equalizer is adjusted using a digital signal processor (hereinafter referred to as a DSP) and a control program therefor. The present invention relates to a computer-readable recording medium.

[0002]

2. Description of the Related Art A conventional effect-adjusting sound apparatus is disclosed, for example, in JP-A-8-32385. This is, as shown in FIG. 9, input means 1 for effect adjustment, and DSP control means 2 for determining and storing coefficients by effect adjustment by the input means 1.
And effect processing means 9 for effecting an input audio signal from an input terminal 7 based on the coefficient and outputting the processed signal to an output terminal 8. Therefore, in the effect-adjusting sound device 6, the DSP
The effect adjustment of the input audio signal can be performed only by transferring the coefficient data from the control means 2.

FIG. 10 is a block diagram showing details of the effect adjusting sound device 6. To state this,
7 is a digital audio signal input terminal, 8 is a digital audio signal output terminal, 15 to 17 are input audio signals, 27 to 29
Is an output audio signal, 18 to 20 are M registers, 24 to
Reference numeral 26 denotes M multipliers, 2 denotes DSP control means for writing coefficients to each of the registers 18 to 20, and 9 denotes effect processing means for effecting an audio signal from the input terminal 7 and outputting an audio signal to the output terminal 8. Output as

Next, the operation will be described. First, when the coefficient data is transmitted from the DSP control means 2, it is stored in the registers 18 to 20, respectively. Next, the input audio signals 15 to 17 are multiplied by coefficient data from the registers 18 to 20 in the corresponding multipliers 24 to 26, and output to the output terminal 8 as output audio signals 27 to 29. As described above, the effect adjustment can be performed by transferring the coefficient data from the DSP control unit to the effect processing unit 9.

However, in such an effect-adjusting sound apparatus, if effect processing with different frequency characteristics or level characteristics is performed successively, the waveform of the output audio signal becomes discontinuous depending on the timing of the effect adjustment, and the output audio signal becomes discontinuous. Noise was mixed into the

The reason for this will be described with reference to FIG. 18 which shows a change in a sound waveform depending on the timing of effect adjustment. In the figure, reference numeral 10 denotes an output audio signal near a specific frequency, and the amplitude on the vertical axis indicates the sound intensity. The adjustment timing 11 indicates the timing at which the effect is adjusted by the input unit 1 in FIG. Here, A12 indicates an adjusted audio waveform generated when coefficient data is set in the registers 18 to 20 from the DSP control means 2 by the effect adjustment in the input means 1 in the effect processing means 9 in FIG. B13 is an ideally adjusted sound waveform that has been subjected to effect processing without causing a sharp change in the waveform at the adjustment timing 11. If the waveform suddenly changes at the adjustment timing 11 like the adjusted audio waveform A12 after the effect adjustment, it is output from the output terminal 8 as an audio signal mixed with noise.

On the other hand, in order to improve such inconvenience, an effect-adjusting sound apparatus as shown in FIG.
This is disclosed in JP-A-8-32385. Describing this, 7 is an input terminal of a digital audio signal, 8 is an output terminal of a digital audio signal, 15 to 17 are input audio signals, 27 to 29 are output audio signals, 18 to 20 are M registers, and 24 to 26 is M multipliers, 2 is DSP control means, 21 to 23 are integrators for smoothing the outputs of the registers 18 to 20 and input to the multipliers 24 to 26, and 36 is effect processing means.

FIG. 12 shows the internal configuration of the integrators 21 to 23, 30 is an input terminal, 31 is a multiplier for multiplying the input signal by a multiplication coefficient (1-α), 35 is an output terminal,
3 is a delay unit for delaying the multiplied output signal, 34 is a multiplier for multiplying the delayed output signal by a multiplication coefficient α, and 32 is a multiplier 3
It is an addition unit that adds each of the output signals 1, 34.

Next, the operation will be described. When the coefficient data is transmitted from the DSP control means 2, the coefficient data is stored in the registers 18 to 20. In addition, these registers 1
The integrators 21 to 23 receive coefficient data of 8 to 20 as inputs.
Smoothes the coefficient data and inputs it to multipliers 24-26. On the other hand, the input audio signal 1 input from the input terminal 7
The multipliers 5 to 17 are multiplied by the coefficient data smoothed by the integrators 21 to 23 in the corresponding multipliers 24 to 26, and output to the output terminal 8 as output audio signals 27 to 29.

On the other hand, each of the integrators 21 to 23 has the same circuit configuration. In the integrators 21 to 23, the input signals stored in the registers 18 to 20 by the DSP control means 2 and input to the input terminal 30 are the same. Multiplying unit 3 multiplies the multiplication coefficient (1−α) by
Multiply by one. The output data of the output terminal 35 is delayed by the delay unit 33, the multiplied unit 34 multiplies the delayed data by a multiplication coefficient α, and the multiplied output is added to the input signal by the adding unit 32.

FIGS. 13 and 14 to 16 are flow charts showing the operation procedure of the input means 1 and the DSP control means 2. In FIG. 13, when the input of the effect adjustment is performed by the input means 1 (step S1), DS
After preparing for P transmission (step S2), the input process is terminated (step S3). Also, as shown in FIG. 14, when the DSP transmission preparation is started (step S4), the DSP
Level ALV adjusted as input data to control means 2
r is set (step S5). The level ALV to be adjusted is, for example, the value of the level number 10 when the level is adjusted from the level 0 setting to the level 10 by the equalizer adjustment. Thereafter, the preparation for DSP transmission ends (step S).
6). Further, as shown in FIG. 15, in the DSP control (step S7), the DSP coefficient is transmitted (step S7).
8) Then, the adjustment ends (step S9). Then, in the DSP coefficient transmission of FIG. 16 (step S10),
The coefficient data to be transmitted to the DSP is obtained from the coefficient data table Ceft selected by the adjustment level ALVr set in the preparation for the DSP transmission (step S11). Next, the coefficient data acquired from the coefficient data table Ceft as shown in FIG. 17 is transmitted to the DSP (Step S).
12) Then, the transmission ends (step S13). In the coefficient data table Ceft, coefficient data for each level to be adjusted is stored in the microcomputer in advance.

[0012]

However, in such a conventional effect adjusting device, it is necessary to add integrators 21 to 23 to the effect processing means 36, and the scale of the effect processing means 36 is large. There was a problem of becoming. This means that the effect processing for realizing the integrators 21 to 23 requires four steps of processing for one integrator, and all the outputs of the registers 18 to 20, which are noise sources, are output to the integrators. Is required, so that the circuit scale becomes large.

The smoothing convergence time of the integrators 21 to 23 is important for calculating the value of the multiplication coefficient α of the multiplication unit 34 constituting the integrators 21 to 23.
The convergence time is increased by smoothing, and if the convergence time is shortened, sufficient smoothing characteristics cannot be obtained, resulting in a problem that noise cannot be prevented.

The reason is that the factors causing the problem can be understood by formulating the integrators 21 to 23 and changing the value of the multiplication coefficient α to perform the calculation. Next, the integrators 21 to
23 is a mathematical expression. Now, let X be the integrators 21 to
23, the coefficient data transmitted from the DSP control means 2 at the input terminal 30 and Yr is the coefficient data integrated by the integrators 21 to 23, that is, the coefficient data of the previous output terminal 35 processed by the integrators 21 to 23. Assuming that the coefficient data is output, Yr = Y = X × (1−α) + Yr × α.

For example, assume that when Yr is 0, 1 is input to X from the input terminal 30. The multiplication factor α is
The multiplication coefficient α is set to 0.8 since the integrator smoothes the data most when the decimal data is close to 1. When these values are given to the mathematical expression, the first mathematical expression calculation result Y, that is, 0.2 is output from the output terminal 35. Further, the result Y of the second mathematical calculation is 0.36. In this way, the process by the integrators 21 to 23 is performed 76 times until the output of the input terminal 30 is finally smoothed to 1. The time for performing the 76th processing is determined by the DSP
For example, in the case of a DSP operating at a sampling frequency of 44.1 KHz, about 1.7
msec time is required.

Further, the multiplication coefficient α is calculated according to Japanese Patent Application Laid-Open No. 8-32385.
Calculated as the value 0.00814 described in the invention described in Japanese Patent Laid-Open Publication No. H10-209, the first mathematical calculation result Y is 0.99186
Becomes From this result, the coefficient data output from the integrators 21 to 23 changes rapidly from 0 to 0.99186, which is not smoothed. Therefore, noise is mixed into the output audio signal from the output terminal 8 and the noise is reduced. The prevention effect cannot be obtained.

The present invention has been made to solve the above-described problems, and can prevent noise from being mixed into output sound due to abrupt level changes due to effect adjustment, particularly to equalizer adjustment, and can reduce the circuit scale. It is an object of the present invention to provide a method of controlling an effect-adjusting sound device which can be realized quickly without increasing the size of the effect adjusting sound device, and a recording medium storing the control program.

[0018]

According to one aspect of the present invention, there is provided a method for controlling an effect-adjusting sound apparatus, comprising the steps of: inputting an effect adjustment by an input means; A digital signal processor transmission preparation step of setting data for deriving coefficient data stored in the microcomputer based on the level, and a digital signal processor control means for converting the coefficient data based on the effect-adjusted level. Determining the coefficient data to be determined; transmitting the determined coefficient data to the effect processing means in a stepwise manner up to the adjusted level; and terminating the effect adjustment by terminating the transmission of the coefficient data corresponding to the adjusted level. Effect adjustment end step In which was to so that.

According to a second aspect of the present invention, in the control method of the effect-adjusting sound apparatus, the effect adjustment is an equalizer adjustment for raising or lowering the level of a specific frequency.

Further, the recording medium according to the third aspect of the present invention allows an effect adjustment to be input, prepares for transmission of coefficient data to the effect processing means, and determines coefficient data to be transmitted to the effect processing means. The control program of the effect-adjusting sound device that causes the coefficient data to be transmitted stepwise to the adjusted level and terminates the transmission of the coefficient data is recorded.

[0021]

DESCRIPTION OF THE PREFERRED EMBODIMENTS Next, an embodiment of the present invention will be described in detail with reference to the flowcharts of FIGS. 1 and 2 and a block diagram of FIG. First, in the input processing, effect adjustment is performed, and in particular, the level of a specific frequency at the time of equalizer adjustment is raised or lowered (step S2).
1). At this time, a check is made by the input means 1 so that the level does not exceed the upper limit and the lower limit. Then DS
Preparation for P transmission is performed (step S22). In this step S22, the coefficient data for transmitting the adjusted level to the effect processing means 9 is stored in the R inside the microcomputer.
This is stored in the AM, thereby ending the input processing (step S23).

Next, in the DSP control means 2, the input means 1
The coefficient data to be transmitted to the effect processing means 9 is determined based on the level adjusted in step (1), and the processing for transmitting the coefficient data is performed (step S24). Next, DSP coefficient transmission is performed (step S25). Here, determination of coefficient data to be transmitted to the effect processing means 9 and transmission processing are performed. By transmitting the coefficient data determined by the DSP coefficient transmission stepwise to the adjusted level, the coefficient data changes to a level having a large difference from the level before adjustment, that is,
Prevents noise input due to sudden input voice waveforms.
Subsequently, in the adjustment end processing (step S26), it is checked whether coefficient data corresponding to the adjusted level has been transmitted, and if not transmitted, repetition coefficient transmission 4 is performed. When the transmission ends, the adjustment ends (step S27).

The above-mentioned DSP transmission preparation processing is performed in accordance with FIG.
When the DSP transmission preparation process is started (step S28), first, the target adjustment level ALV
x is substituted for the level ALVr to be adjusted (step S2)
9). Here, the level indicates the smallest unit that can be adjusted. To adjust the level in the plus / minus direction,
Currently sent adjustment level ALV and adjustment level AL
The magnitude of Vr is compared. If the level ALVr to be adjusted is large, a plus value is set. If the level is small, a minus value is set (step S30), and the DSP transmission preparation is terminated (step S31).

Next, the above-described DSP coefficient transmission processing is performed in accordance with FIG.
When the DSP coefficient transmission process is started (step S32), the level LVdr is checked for plus and minus (step S33), and in the case of the plus direction, 1 is added to the adjustment level ALV currently transmitted. Add (step S34). In the case of the minus direction, 1 is subtracted from the currently transmitted adjustment level ALV (step S35). As a result, one level of the coefficient data corresponding to the adjustment level ALV currently transmitted is
The level of the coefficient data in the plus direction or the minus direction is set to the adjustment level ALV that has been currently transmitted. Next, coefficient data to be transmitted is obtained from the coefficient data table Ceft (step S36), and this is
The data is transmitted to the SP (step S37), and the process ends (step S38).

The process for terminating the transmission is as shown in FIG. 5, and when this process is started (step S3).
9), target adjustment level ALV set in step S30
x is compared with the currently adjusted adjustment level ALV (step S40), and if they are the same level (step S4).
1), the adjustment is completed (step S42), and if the levels are not the same, the DSP coefficients are transmitted from step S32 onward (step S43). For example, the currently transmitted adjustment level ALV is 0, and the adjustment level ALVr is 1
If it is set to 5, the target adjustment level ALVx becomes 15 in step S30. In step S33, the level LVdr in the plus / minus direction becomes plus. As described above, since LVdr is positive in the DSP coefficient transmission processing, 1 is added to ALV in step S34, and ALV becomes 1. In step S36, AL
The coefficient data of level 1 by V is a coefficient data table C
Then, the level 1 coefficient data is transmitted to the effect processing means 9 in the processing of step S37. Further, in the adjustment end processing, ALVx (= 15) and ALVx (= 15)
(= 1), the result is not the same level,
The coefficient is transmitted (step S43). In this way, the ALV transmits coefficient data to the effect processing means 9 step by step at every level in the order of 1 → 2 → 3 until the level becomes the same.

Next, another embodiment of the present invention will be described with reference to FIGS.
This will be described with reference to the flowcharts of FIG. First, in FIG. 6, when the DSP transmission preparation processing is started (step S44), the difference between the currently transmitted adjustment level ALV and the level ALVr to be adjusted is calculated as an absolute value, and this is added to the addition / subtraction adjustment level data storage table. It is stored in ADlt as data LVof serving as an offset (step S45). The level plus or minus LVdr is the currently transmitted adjustment level ALV and the level ALV to be adjusted
Level ALV to compare r, check magnitude relationship and adjust
If r is large, a plus is set, and if r is small, a minus is set (step S46). Thereafter, the position Tp for acquiring the data of the table is cleared to 0 (step S47), and the process of preparing for DSP transmission is ended. (Step S48).

On the other hand, as shown in FIG. 8, when the transmission of the DSP coefficient is started (step S49), the data sequence in which the addition / subtraction adjustment level data ADld is transmitted to the DSP has a sequence for each LVof. The column is selected from the addition / subtraction adjustment level data storage table ADlt by LVof, and ADld for addition or subtraction to the currently transmitted adjustment level is acquired by Tp (step S5).
0). Then, 1 is set to Tp to acquire the next ADld.
Is added (step S51), and the process of transmitting the DSP coefficient ends (step S52).

As shown in FIG. 7, when the adjustment end process is started (step S53), it is checked whether or not ADld is an end code which is data other than the addition / subtraction adjustment level data (step S54). If so, the adjustment ends (step S55). On the other hand, AD
If ld is not the end code, from the set LVdr,
If it is positive, ADld is added to ALV, and if it is negative, ADld is subtracted from ALV (step S5).
6). Subsequently, coefficient data to be transmitted to the DSP is obtained from the coefficient data table Ceft of FIG. 17 selected by the calculated ALV (step S57).

Then, the adjustment data is transmitted to the obtained coefficient data DSP (step S58), and the adjustment processing is terminated (step S59). After this transmission, D in step S49
The process proceeds to SP coefficient transmission. According to this embodiment, the same noise prevention effect as that of the above-described embodiment can be obtained. In the case of the adjustment from the minimum adjustment level to the maximum adjustment level, in the above-described embodiment, the DSP is set for each adjustment level. However, in this embodiment, the processing speed of the microcomputer can be increased by setting the data of the addition / subtraction adjustment level data storage table ADlt so as to have an optimum adjustment level width.

[0030]

As described above, according to the present invention, the equalizer adjustment, which is the effect adjustment, is performed by shifting the coefficient data one level higher or lower than the coefficient data for which the coefficient data to be transmitted to the effect processing means is already set. Since the transmission is performed stepwise, it is possible to reliably prevent the waveform of the frequency audio signal from becoming discontinuous due to the timing of effect adjustment, that is, to prevent noise from occurring due to a sudden change in coefficient data. . For example, in the case of adjustment of 1 dB per level, coefficient data equivalent to 1 dB is transmitted to the effect processing means step by step to the target level for each level, so that generation of noise can be prevented. Further, as described above, since the coefficient data is transmitted to the effect processing means in a stepwise manner, an integrator is not required, and the effect that the circuit scale can be reduced can be obtained.

Further, according to the present invention, since an integrator is not required, the convergence time is increased by smoothing the coefficient data, and the noise which cannot be smoothed is reduced by shortening the smoothing convergence time. This can eliminate the adverse effect of inviting. The reason is that there is always an upper limit and a lower limit for level adjustment in equalizer adjustment. Therefore, smoothing can be realized by dividing the lower limit level into several levels from the upper limit, and noise data is not generated by one level adjustment. Corresponds to each level.

[Brief description of the drawings]

FIG. 1 is a flowchart showing an effect adjustment input processing procedure in a control method of an effect adjustment sound device according to an embodiment of the present invention.

FIG. 2 is a flowchart showing a digital signal processor control processing procedure according to the present invention.

FIG. 3 is a flowchart showing a digital signal processor transmission preparation processing procedure according to the present invention.

FIG. 4 is a flowchart showing a digital signal processor coefficient transmission processing procedure according to the present invention.

FIG. 5 is a flowchart showing an adjustment end processing procedure according to the present invention;

FIG. 6 is a flowchart showing a digital signal processor transmission preparation processing procedure in the control method of the effect-adjusting sound device according to another embodiment of the present invention.

FIG. 7 is a flowchart showing another adjustment end processing procedure according to the present invention.

FIG. 8 is a flowchart showing another digital signal processor coefficient transmission processing procedure in the present invention.

FIG. 9 is a block diagram showing a general effect-adjusting sound device.

FIG. 10 is a block diagram showing details of a conventional effect-adjusting sound device.

FIG. 11 is a block diagram showing another conventional effect-adjusting sound device.

FIG. 12 is a block diagram showing details of an integrator in FIG. 11;

FIG. 13 is a flowchart illustrating a conventional effect adjustment input processing procedure.

FIG. 14 is a flowchart showing a conventional digital signal processor transmission preparation procedure.

FIG. 15 is a flowchart showing a conventional digital signal processor control processing procedure.

FIG. 16 is a flowchart showing a conventional digital signal coefficient transmission processing procedure.

FIG. 17 is an explanatory diagram showing a conventional coefficient data table.

FIG. 18 is an explanatory diagram showing a change in an output audio waveform according to a conventional effect adjustment timing.

[Explanation of symbols]

DESCRIPTION OF SYMBOLS 1 Input means 2 DSP control means (digital signal processor control means) 6 Effect adjustment sound device 9 Effect processing means

Claims (3)

[Claims] 1. A control method for an effect-adjusting audio apparatus in which a microcomputer controls a digital signal processor according to a control program, comprising: an input processing step of inputting an effect adjustment by an input unit; A digital signal processor transmission preparation step of setting data for deriving coefficient data stored in the computer; and determining coefficient data based on the effect-adjusted level by the digital signal processor control means. Transmitting the determined coefficient data to the effect processing means in a stepwise manner up to the adjusted level; and terminating the effect adjustment by terminating the transmission of the coefficient data corresponding to the adjusted level. The method of the effect adjustment sound apparatus and executes an-object adjustment end step. 2. The control method according to claim 1, wherein the effect adjustment is an equalizer adjustment for raising and lowering a level of a specific frequency. 3. A recording medium in which a control program of an effect adjustment sound device for controlling a digital signal processor is recorded by a microcomputer, the effect adjustment is input, and preparations are made to transmit coefficient data to the effect processing means. A coefficient program to be transmitted to the effect processing means, a stepwise transmission of the coefficient data to an adjusted level, and a termination of the transmission of the coefficient data; Recording medium on which is recorded.