JPH01176104A - Audio equipment - Google Patents

Abstract

PURPOSE:To suppress unpleasant pop noise at changeover of a sound source by varying gradually the frequency characteristic and sound volume at the on/off state of a mute signal for the muting processing (audio erasure processing). CONSTITUTION:A transient filter 17 has a function suppressing selectively a low sound frequency, a medium sound frequency and a high sound frequency in response to a coefficient set by a filter coefficient setting means 18. The filter coefficient setting means 18 set the filter coefficient so as to intensify the selective suppression gradually around the middle sound frequency of an audible signal when a mute signal from a mute signal generating means 14 is switched from OFF to ON state and so as to arise in the opposite direction when switched from ON to OFF state. Thus, the production of pop noise at mode changeover at power on/off is suppressed efficiently.

Description

[Detailed Description of the Invention] [Summary] To suppress unpleasant popping sounds in audio equipment when power is turned on and off, or when switching sound sources such as an AM tuner, FM tuner, compact disc player, or cassette deck. , the frequency characteristics and volume are gradually changed when a mute signal for muting processing (sound erasure processing) provided in a conventional audio device is turned on and off.

[Industrial Field of Application] The present invention relates to an audio device, and particularly to an audio device having a digital signal processing section.

In recent years, DSP (Digital
Products based on digital signal processing using an IT-al Signal Processor have begun to appear on the market. This digital signal processing has advantages such as high precision, no need for adjustment, no deterioration over time, and flexibility, so it is expected to become more popular in the future.

[Conventional technology]

In audio equipment, the signal level may suddenly change during transition periods such as when power is turned on or off, or when switching modes in audio equipment with multiple functions (FM/AM tuner, compact disc player, cassette deck). This causes an unpleasant popping sound.

In audio devices that employ digital signal processing using the DSP described above, the popping sounds during these transitional periods have become even more noticeable. In order to suppress this, muting processing has been conventionally performed to cut off the signal before turning on/off the power supply or switching the mode.

FIG. 6 shows a typical example of a muting circuit employed in a conventional audio device. In this figure, the contact 51 of the reed relay is normally closed, and a signal is supplied from the speaker drive means 12 to the speaker 13. During the transition period, the mute signal output from the mute signal generating means 14 opens the contact 51, thereby cutting off the signal. Rapid changes in the signal level due to the opening and closing of the contact 51 are alleviated by the action of the resistor R and the capacitor C, and the popping noise is suppressed.

(Problem to be solved by the invention) In the above-mentioned type of muting circuit, a CR circuit is inserted into the signal path, so these frequency characteristics degrade the characteristics of the entire audio device. .

Therefore, an object of the present invention is to provide an audio device using digital signal processing that eliminates such drawbacks.

[Means for solving problems]

FIG. 1 is a diagram showing the principle of the audio device of the present invention. The audible signal from the sound source circuit selected by the sound source switching means 11 is passed through the transient filter 17 to the speaker driving means 12.
and drives the speaker 13. The filter coefficients of the transient filter 17 are given from the filter coefficient setting means 18. The mute signal generating means 14 generates a mute signal during transition periods such as before and after the power is turned on and off, and before and after switching by the sound source switching means 11, and transmits it to the filter coefficient setting means 18.

The above-mentioned transition period filter 17 has a filter coefficient setting means 18.
It has a function of selectively suppressing the bass range, middle range, and phonograph range of the audible signal according to a coefficient set from . When the mute signal from the mute signal generating means 14 is switched from OFF to ON, the filter coefficient setting means 18 sequentially sets the filter coefficients so as to gradually strengthen selective suppression centered on the middle range of the audible signal, When switching from on to off, set the filter coefficient so that it changes in the opposite direction.

[Function] When the mute signal changes from off to on, it suppresses changes in the low and high ranges, which are the areas where noise occurs, and gradually strengthens only the suppression of the midrange. After sufficient suppression, the low and high ranges change. However, in the transition period from off to on, the opposite change occurs, so unpleasant popping noises are suppressed.

〔Example〕

FIG. 2 is a block diagram showing the hardware configuration of a specific example of the audio device of the present invention.

Those corresponding to the sound source circuits (16-1 to 16-n) in FIG. The audible signal output is once converted into a digital signal by the A/D converter 21, processed by the DSP 22, and then converted again into an analog signal by the D/A converter 23 and supplied to the power amplifier 121. The A/D conversion period is set to a sampling period (for example, 25μ5ec) that can process up to the upper limit of the audible signal frequency (for example, 20kHz), and therefore the processing within the DSP 22 is designed to be able to process within this sampling period. has been done. For example, the multiplier 30 and adder 31 in the DSP 22 are connected by a pipeline, and perform a product-sum operation (aIXx, + az XXz '+ . . 'a
A set of multiplications and additions in units of 100n
It is possible to execute the command once every microinstruction execution cycle of about s. Input/output of digital signals and input of a mute signal from the control microcomputer 141 are performed via an input/output interface (Ilo) 26.

Based on such a configuration, the transition period filter 17 in FIG.
and the filter coefficient setting means 18 are realized by the hardware and software (described later) of the DSP 22.

In addition, the DSP 22 also has sound quality and volume processing functions using digital processing, similar to conventional audio devices.

The control microcomputer 141 is similar to the control microcomputer in a conventional audio device with a built-in microcomputer, and uses known technology to control the operation before and after power on/off, before and after mode switching, and during automatic or semi-automatic channel selection. Output mute signal. In this way, the control microcomputer 141
It realizes the mute signal generating means 14 shown in the figure, and also performs functions such as automatic or semi-automatic channel selection according to key operations.

FIG. 3 shows that in the specific example described above, the first
FIG. 2 is a diagram showing software processing for realizing the function of the transition period filter 17 shown in the figure in the form of a signal diagram. The multiplication coefficients of the multipliers 211.212.213.214°215, 216 are KLO, KLI, KL2°KL3.
KL4. -1, 217.218 is a delay device, 219
, 220.221 are adders. The path from multiplier 212 to multiplier 215 constitutes a low-pass filter, the path including multiplier 214 constitutes a through-pass, and the path including multiplier 213 constitutes a high-pass filter. The outputs of these three sources are combined by an adder 221, resulting in a result y(n) and an input signal x(n
) is expressed by the difference equation % formula %. From this, the transfer function H(
z) is +KL2 (1-z-') (2
). Therefore, the maximum transmissibility GL, GM, and GH in the low, mid, and high frequencies are GL = K L 3 + K L 4
(3) G M = K L 3
(4) GH=KL3+2・KL2
(5) becomes. However, KLO, KLI
is, K L 1 = KL2 /KL4
(6) K L O = 1- K L 1
It shall be set under the conditions of (7). The following table shows the filter coefficients KL2 for realizing various combinations of GL, GM, and GH calculated based on equations (3), (4), and (5). KL3. The value of KL4 is shown.

KLO and KLl can be calculated from equations (6) and (7).

FIG. 4 shows the frequency characteristics of each filter coefficient in the table, and F1-6 in the figure correspond to filters Nα1-6 in the table. As is clear from the figure, filter Nα1 in the table
It can be seen that as the filter coefficients corresponding to Nα are sequentially switched to Nα2 to Nα6, the region centered on the midrange (1000) Iz) is selectively and gradually suppressed.

FIG. 5 is an example of a flowchart for realizing the SP software as described above. To explain sequentially with reference to this figure, when the program starts, initial processing such as memory initialization (for example, filter No. initialization) and input/output port initialization is performed (step a). Thereafter, the main routine repeats the idle process (step b). When a sampling timing interrupt of a fixed period occurs, the state of the mute signal from the control microcomputer is first input (step C).
If it is on and the filter Nα is not 6 (steps d and e), the filter No. is incremented. If it is off and the filter number is 1 (step d,
g) Decrement the filter number. Next, A/
The conversion result by the D converter 21 is taken in from the input/output port and used as the current value of the discrete signal x (n) (step i)
. If the filter Nα is 1 (step j), normal calculations for sound quality and volume processing are performed (step k), and the results are output (step n). If the filter Nα is not 1, set the filter coefficient corresponding to the filter number based on the values in the table.Step 1) 1. (n) value, X
Previous value x(n-1) of (n), previous value y(n) of y(n)
), y(n) is calculated based on equation (1) (step m) and output (step n).

As mentioned above, each time an interrupt occurs at a certain period, step c
By executing the process -n, the transition period filter 17 and filter coefficient setting means 18 in FIG. 1 are realized.

The frequency characteristics shown in FIG. 4 are similar to those of an automatic hearing correction circuit (automatic loudness circuit) that emphasizes high and low frequency ranges while decreasing the volume.

Therefore, by using the filter coefficients and arithmetic routine of the automatic loudness circuit, the audio device of the present invention can be realized without the need to newly add these.

〔Effect of the invention〕

As described above, the audio device using the digital signal processing of the present invention can effectively suppress the occurrence of popping sounds when turning the power on and off and when switching modes.
Moreover, since the filter coefficients and calculation routines used in conventional automatic loudness circuits can be used as they are, the amount of software does not increase significantly.

[Brief explanation of the drawing]

FIG. 1 is a principle diagram of the audio device of the present invention, FIG. 2 is a block diagram showing the hardware configuration of a specific example of the audio device of the present invention, and FIG. 3 is a block diagram of a specific example of the audio device of the present invention. FIG. 4 is a signal diagram showing the filter processing of the DSP software. FIG. 4 is a diagram showing the frequency characteristics of the filter coefficients in the table. FIG.
FIG. 6 is a flowchart showing SP software processing, and FIG. 6 is a diagram showing the configuration of a conventional muting circuit. In the figure, 11...... Sound source switching means, 12...
......Speaker driving means, 14...
......Mute signal generating means, 16-1 to 1
6-n...Sound source circuit, 17...Transition period filter, 18.
......Filter coefficient setting means, 22.
・・・・・・・・・・・・DSP. Figure 3 shows the configuration of a conventional muting circuit, showing the configuration of a conventional muting circuit @6 times

Claims (1)

[Claims] 1. One or more sound source circuits (16-1 to 16-n), and amplifying an audible signal from one of the sound source circuits (16-1 to 16-n) to a speaker ( 13); and a mute signal generating means (14) that detects a transition period of the audible signal and outputs a mute signal ON. a transition period filter (17) that selectively suppresses the bass, midrange, and treble ranges of the audible signal, and a mute signal from the mute signal generating means (14) that is switched from off to on in response to When the mute signal is switched from on to off, the filter coefficients of the transient filter (17) are sequentially set so as to gradually strengthen the selective suppression mainly on the midrange of the audible signal, and when the mute signal is switched from on to off, An audio device comprising filter coefficient setting means (18) for setting the filter coefficient so as to change in the opposite direction. 2. The audio device according to claim 1, wherein the mute signal generating means (14) turns on the mute signal during a period before and after the power of the audio device is turned on and off. 3. Selecting any one of the one or more sound source circuits (16-1 to 16-n) to drive the speaker (12)
Claim 1, further comprising a sound source switching means (11) connected to the sound source switching means (11), wherein the mute signal generating means (14) turns on the mute signal during a period before and after the sound source switching means (11) switches the connection. The audio device according to item 1 or 2. 4. At least one of the sound source circuits (16-1 to 16-n) is a tuner that tunes and detects and amplifies radio waves of a specific frequency, and the mute signal generating means (14) detects and amplifies radio waves of a specific frequency. Claims 1 to 3 which are turned on when the tuner is supplied to the speaker driving means (12), when the tuning frequency of the tuner is changed, and turned off when the tuner is tuned. The audio device according to any one of . 5. The audio device according to any one of claims 1 to 4, wherein the transition period filter (17) is a digital filter.