JP2008153999A - Audio processing circuit, starting method thereof and electronic apparatus using these - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To solve the problem that pop-up noise occurs because there is a lower limit in pulsewidth of an output pulse of a ΔΣ modulator. <P>SOLUTION: In a digital audio processing circuit 100, a transition signal generating part 22 generates a transition signal upon starting the circuit. The ΔΣ modulator 40 performs ΔΣ modulation of the transition signal S7. A duty ratio adjusting part 60 adjusts a duty ratio of a bit stream S6 output from the ΔΣ modulator 40 and outputs the duty ratio to a class D amplifier 50 of a post stage. The duty ratio adjusting part 60 changes the pulsewidth of the bit stream S6 to thereby adjust the duty ratio. The duty ratio adjusting part 60 gradually increases the pulsewidth of the bit stream S6 from a predetermined minimum value to the pulsewidth of an input bit stream with the elapse of time upon starting the circuit. <P>COPYRIGHT: (C)2008,JPO&INPIT

Description

  The present invention relates to audio signal processing, and more particularly to a 1-bit digital analog (D / A) converter.

  With the development of semiconductor integrated technology in recent years, 1-bit D / A conversion using high-speed digital signal processing is performed in electronic devices having an audio reproduction function such as a silicon audio player, a CD (Compact Disc) player, and a mobile phone terminal. Used. In 1-bit D / A conversion, an audio signal is first oversampled using a digital filter to remove unnecessary bands. Subsequently, the filtered audio signal is converted into a 1-bit pulse signal that is pulse-modulated using a ΔΣ modulator or the like. Subsequently, the pulse signal is amplified using a class D amplifier, and high frequency components are removed by an analog filter. The resulting analog filter output signal is the audio signal to be reproduced.

When the device is activated, noise called pop-up noise is generated when the voltage applied to the speaker, headphones, or the like changes abruptly. In order to suppress pop-up noise, it is necessary to gradually increase the voltage applied to the speaker.
JP 2003-318664 A

  The ΔΣ modulator performs modulation processing using a clock signal having a predetermined frequency. Therefore, the minimum value of the duty ratio (for example, pulse width) of the pulse signal output from the ΔΣ modulator is limited by the clock signal. Therefore, even if the input signal to the ΔΣ modulator is gently changed, the continuity of the analog audio signal obtained by smoothing the generated pulse signal is impaired, and there is a problem that pop-up noise occurs.

  The present invention has been made in view of these problems, and a comprehensive object thereof is to provide an audio processing technique with reduced pop-up noise.

  An audio processing circuit according to an aspect of the present invention includes a transition signal generation unit that generates a transition signal when the circuit is activated, a ΔΣ modulator that ΔΣ modulates the transition signal, and a duty ratio of a bit stream output from the ΔΣ modulator. And a duty ratio adjustment unit that outputs to a subsequent class D amplifier.

  According to this aspect, the duty ratio adjustment unit is provided, and a pulse signal having a duty ratio lower than that of the bit stream generated by the ΔΣ modulator is generated, so that an audio output unit (electroacoustic conversion element) such as a speaker or headphones can be used. Since the supplied drive voltage can be continuously increased from around 0 V, it is possible to prevent the voltage from being discontinuously changed and to suppress pop-up noise.

  Note that “duty ratio” means the ratio of a high-level period to the cycle time of the bit stream, and in this specification, not only the duty ratio (on time / cycle time) for a single pulse but also a plurality of pulses. It may mean the average value of the duty ratio. Therefore, “adjusting the duty ratio” includes changing the pulse width and changing the appearance frequency of the pulses.

  The duty ratio adjusting unit may adjust the duty ratio by changing a pulse width of the bit stream.

The duty ratio adjustment unit may gradually increase the pulse width of the bit stream from a predetermined minimum value to the pulse width of the input bit stream as time elapses when the circuit is activated.
In this case, since the drive voltage for the electroacoustic transducer can be changed gradually, pop-up noise can be further suppressed.

The duty ratio adjusting unit may receive a second clock signal having a higher frequency than the first clock signal supplied to the ΔΣ modulator, and adjust the pulse width of the bit stream using the second clock signal. .
Since the pulse width of the second clock signal is narrower than the pulse width of the first clock signal, a bit stream having a narrower pulse width than the bit stream generated by the ΔΣ modulator can be obtained by using the second clock signal. Can be generated.

The frequency of the first clock signal is m times the sampling frequency of the input audio signal (m is a natural number), and the frequency of the second clock signal is n times the sampling frequency of the input audio signal (n is natural number satisfying n> m).
In this case, since the first and second clocks can be generated by multiplying the same master clock signal, the circuit configuration can be simplified.

The duty ratio adjusting unit may change the density by thinning out pulses from the bit stream at a predetermined rate.
In addition to adjusting the pulse width, the pulse stream duty ratio can be set lower by thinning out the pulses. Further, the duty ratio can be set low by performing the process of thinning out pulses independently.

  The audio processing circuit may further include a selector circuit that receives the audio signal to be reproduced and the transition signal generated by the transition signal generation unit, selects one of them, and outputs it to ΔΣ modulation.

The transition signal generation unit fixes the transition signal at a predetermined level for a predetermined period from the start and then increases to the midpoint level with time. The duty ratio adjustment unit performs the pulse width of the bit stream during the period when the transition signal is at the predetermined level. May be set to an active state in which the bit stream is output and the bit stream may be output as it is during a period in which the level of the transition signal increases.
According to this configuration, it is possible to increase the duty ratio by increasing the duty ratio by the duty ratio adjusting unit for a predetermined period and then increasing the transition signal.

  The predetermined level may be equal to or lower than a level at which the pulse width of the bit stream generated by the subsequent ΔΣ modulator is minimized.

  The audio signal processing circuit may be integrated on a single semiconductor substrate. “Integrated integration” includes the case where all of the circuit components are formed on a semiconductor substrate and the case where the main components of the circuit are integrated. A resistor, a capacitor, or the like may be provided outside the semiconductor substrate. By integrating the circuit as one IC, the area can be reduced.

  Another embodiment of the present invention is an electronic device. This electronic device includes a signal generation unit that generates a digital audio signal, the above-described audio processing circuit that receives the audio signal as an input, a class D amplifier that amplifies the bit stream from the duty ratio adjustment unit of the audio processing circuit, A filter for filtering the output signal of the class D amplifier, and an audio output unit provided at the subsequent stage of the filter.

  According to this aspect, the pop-up noise generated from the audio output unit can be suppressed, and the added value of the electronic device can be increased.

  Yet another embodiment of the present invention relates to a method for starting a digital audio processing circuit having a ΔΣ modulator. The method includes a step of generating a transition signal when the circuit is activated, a step of ΔΣ modulation of the transition signal, and a step of adjusting a duty ratio of a bit stream generated by the ΔΣ modulation to increase with time. .

  It should be noted that any combination of the above-described constituent elements and a representation obtained by converting the expression of the present invention between methods, apparatuses, and the like are also effective as an aspect of the present invention.

  According to the present invention, pop-up noise can be reduced.

  The present invention will be described below based on preferred embodiments with reference to the drawings. The same or equivalent components, members, and processes shown in the drawings are denoted by the same reference numerals, and repeated descriptions are omitted as appropriate. The embodiments do not limit the invention but are exemplifications, and all features and combinations thereof described in the embodiments are not necessarily essential to the invention.

  FIG. 1 is a block diagram showing a configuration of an electronic device 200 equipped with a digital audio processing circuit 100 according to an embodiment of the present invention. The electronic device 200 is a device that can output sound from a speaker, a headphone, an earphone, or the like such as a mobile phone terminal, a silicon audio player, or a CD player. The electronic device 200 includes a digital audio processing circuit 100, a low-pass filter 110, an audio output unit 120, and an audio signal generation unit 130.

  The digital audio processing circuit 100 and the audio signal generation unit 130 are supplied with a battery voltage or a voltage obtained by stabilizing the battery voltage with a switching regulator or the like as the power supply voltage Vcc. The audio output unit 120 is a device that converts an electrical signal into an acoustic wave, such as a speaker, headphones, or earphones, and is built in or externally attached to the electronic apparatus 200.

  The audio signal generation unit 130 generates a digital audio signal S1. The audio signal S1 is a signal obtained by decoding an audio signal that has been encoded and recorded in a voice, ringtone, or a memory (not shown) spoken by the other party.

The digital audio processing circuit 100 and the low-pass filter 110 in FIG. 1 function as a 1-bit D / A converter.
The audio signal generation unit 130 and the digital audio processing circuit 100 are connected via a signal line 132. For example, the signal line 132 is an I2S standard bus, and the audio signal S1 is transmitted to the digital audio processing circuit 100 as serial data. Note that I2S is merely an example, and other serial buses or parallel buses may be used. The present invention is not limited to a situation where a specific bus is used.

  The digital audio processing circuit 100 receives the audio signal S1, converts it into a pulse-modulated 1-bit pulse signal, amplifies it, and outputs it to the low-pass filter 110 at the subsequent stage. The low-pass filter 110 removes a high frequency component of the output pulse (hereinafter referred to as an output audio signal S2) of the digital audio processing circuit 100 and converts it into an analog audio signal S3. The audio output unit 120 is driven by the analog audio signal S3 from the low pass filter 110.

The digital audio processing circuit 100 includes an input terminal 102 and an output terminal 104. The audio signal S1 from the audio signal generator 130 is input to the input terminal 102, and the output terminal 104 is connected to the low-pass filter 110.
The digital audio processing circuit 100 includes an input interface unit 10, a digital interpolation filter (hereinafter simply referred to as an interpolation filter) 20, a transition signal generation unit 22, a selector 24, a control unit 30, a ΔΣ modulator 40, a class D amplifier 50, a duty ratio. The adjustment unit 60 is included and integrated on a single semiconductor substrate.

  The control unit 30 is a block that comprehensively controls the digital audio processing circuit 100. Specifically, the control unit 30 controls operations of the transition signal generation unit 22, the selector 24, the ΔΣ modulator 40, and the duty ratio adjustment unit 60. A power-on signal PWR_ON is input to the control unit 30 from the outside. When the level of the power-on signal PWR_ON transitions, the control unit 30 controls the first control signal Sc1 to the fourth control signal Sc4 for each circuit block according to a predetermined activation sequence.

  The input interface unit 10 receives the audio signal S1 input via the signal line 132, and serial-parallel converts this to generate an audio signal S4. The audio signal S4 is input to the interpolation filter 20 at the subsequent stage.

  The interpolation filter 20 is a FIR (Finite Impulse Response) filter that interpolates by oversampling the audio signal S4 having the sampling frequency fs by 8 times.

  The transition signal generation unit 22 generates a transition signal S7 that is used when the digital audio processing circuit 100 is activated. The transition signal S7 is a signal whose signal level gradually changes with time, and is used for transition from the mute state to the sound reproduction state (ringing state) (or the reverse transition). When the transition signal generator 22 receives an instruction from the first control signal Sc1 from the controller 30, the transition signal generator 22 starts generating the transition signal S7.

  The transition signal S7 has a waveform that gradually rises from the ground level (digital value corresponding to 0V) to the midpoint level of the audio signal (digital value corresponding to Vcc / 2) in the transition from the mute state to the sound reproduction state. It is preferable to have. Conversely, in the case of transition from the audio playback state to the mute state, it is preferable to have a waveform that gradually falls from the midpoint level of the audio signal to the ground level.

  The selector 24 receives the audio signal filtered by the interpolation filter 20 (hereinafter referred to as an interpolated audio signal S5) and the transition signal S7 generated by the transition signal generation unit 22. The transition signal generator 22 selects and outputs either the interpolated audio signal S5 or the transition signal S7 based on the signal level of the first control signal Sc1. In the present embodiment, the transition signal S7 is selected when the first control signal Sc1 is at a low level, and the interpolated audio signal S5 is selected when the first control signal Sc1 is at a high level.

  The output signal S8 of the selector 24 is input to the subsequent ΔΣ modulator 40. The ΔΣ modulator 40 performs ΔΣ modulation on the signal S8 based on the first clock signal CK1, and outputs a pulse-modulated bit stream (hereinafter referred to as a pulse audio signal S6). The density of the pulse train included in the pulse audio signal S6 or the width (duty ratio) of each pulse corresponds to the amplitude of the audio signal to be reproduced. The ΔΣ modulator 40 may use a generally used higher-order ΔΣ modulator.

  In the present embodiment, the order of the ΔΣ modulator 40 may be switchable. The ΔΣ modulator 40 includes two ΔΣ modulators having different orders, and these may be switched and used, or the order of a single ΔΣ modulator may be switched. The ΔΣ modulator 40 switches the order based on the third control signal Sc3.

  The duty ratio adjusting unit 60 receives the pulse audio signal S6 and the second clock signal CK2. The frequency of the second clock signal CK2 is set higher than the frequency of the first clock signal CK1. The duty ratio adjusting unit 60 adjusts the duty ratio of the pulse audio signal S6 using the second clock signal CK2 when the sound reproduction state is changed from the mute state, and outputs the adjusted signal to the subsequent class D amplifier. The duty ratio adjusting unit 60 adjusts the duty ratio by forcibly changing either one or both of the pulse width and the pulse density. The processing of the duty ratio adjusting unit 60 will be described later.

  The class D amplifier 50 amplifies the pulse signal S9 output from the duty ratio adjustment unit 60. This class D amplifier 50 is a CMOS (Complementary Metal Oxide Semiconductor) inverter type switching amplifier provided between the power supply voltage Vcc and the ground potential. The 1-bit output audio signal S2 amplified by the class D amplifier 50 is amplified to Vcc and output via the output terminal 104.

Next, the adjustment of the pulse audio signal S6 by the duty ratio adjustment unit 60 and the overall operation of the digital audio processing circuit 100 will be described.
FIG. 2 is a time chart showing an operation state at the start-up of the digital audio processing circuit 100 of FIG.
Prior to time t0, the digital audio processing circuit 100 is in a mute state. At time t0, the power-on signal PWR_ON changes from low level to high level, and activation of the digital audio processing circuit 100 is instructed. When the power-on signal PWR_ON becomes high level, the control unit 30 controls the first control signal Sc1 to the fourth control signal Sc4 according to a predetermined activation sequence.

At time t0, the control unit 30 has the second control signal Sc2 at the low level, and the selector 24 selects the transition signal S7.
When the power-on signal PWR_ON becomes high level, the control unit 30 outputs a first control signal Sc1 (not shown) to the transition signal generation unit 22 at the same time or after a predetermined time has elapsed, and the transition signal S7. Instructs generation start of. The transition signal generator 22 fixes the transition signal S7 at the predetermined level L1 or lower after time t0. The predetermined level may be a level at which the pulse width of the pulse audio signal S6 modulated by the subsequent ΔΣ modulator 40 is minimized. After time t0, the transition signal S7 is input to the ΔΣ modulator 40 as the signal S8. When the power-on signal PWR_ON becomes high level, the input audio signal is muted and the interpolated audio signal S5 is fixed at the midpoint level.

  At time t1, the control unit 30 instructs the ΔΣ modulator 40 to start ΔΣ modulation. At this time, the third control signal Sc3 is at a low level, and the order of ΔΣ modulation is set to the second order. After time t1, while the transition signal S7 is at a predetermined level, the duty ratio (pulse width) of the pulse audio signal S6 (not shown) becomes the minimum value. The frequency of the pulse audio signal S6 is defined by the first clock signal CK1, and is set to m times (m is a natural number) the sampling frequency fs of the input audio signal, for example. For example, you may set to m = 128-32.

The duty ratio adjusting unit 60 receives the pulse audio signal S6 in which the duty ratio is set to the minimum value. The control unit 30 instructs the duty ratio adjustment unit 60 to start the duty ratio adjustment process of the pulse audio signal S6 at time t1.
FIG. 3 is a time chart showing the adjustment processing of the duty ratio of the pulse audio signal S6 by the duty ratio adjustment unit 60.

The duty ratio adjusting unit 60 adjusts the duty ratio by executing at least one or both of the following two processes.
1. First Process The duty ratio adjusting unit 60 adjusts the duty ratio by changing the pulse width of the bit stream. Specifically, when the circuit is activated, the pulse width of the bit stream may be gradually increased from a predetermined minimum value to the pulse width of the input bit stream as time passes.

  In the circuit of FIG. 1, the duty ratio adjustment unit 60 receives a second clock signal CK2 having a higher frequency than the first clock signal CK1 supplied to the ΔΣ modulator 40. The duty ratio adjusting unit 60 adjusts the pulse width of the bit stream using the second clock signal CK2. The frequency of the first clock signal CK1 is m times the sampling frequency of the input audio signal, and the frequency of the second clock signal CK2 is n times the sampling frequency of the input audio signal (n is n> m). Natural number). For m = 128 to 32, n = 256 may be set.

  The second clock signal CK2 has a higher frequency than the pulse audio signal S6. Therefore, the duty ratio adjusting unit 60 generates a pulse signal S9 having a pulse width proportional to the number of pulses of the second clock signal CK2 using a counter or a flip-flop.

2. Second Process The duty ratio adjusting unit 60 changes the density by thinning out pulses from the bit stream at a predetermined rate.

  By an arbitrary combination of the first and second processes, the duty ratio adjusting unit 60 gradually changes the duty ratio of the pulse signal S9 from a state close to 0 to approach the duty ratio of the pulse audio signal S6. .

  The duty ratio adjustment unit 60 is set to an active state in which the pulse width of the bit stream is changed during a period in which the transition signal S7 is at a predetermined level (before time t2 in FIG. 2), and a period in which the level of the transition signal S7 increases (FIG. 2). After time t2), the inactive state is set in which the bit stream is output as it is.

  FIG. 3 shows the second clock signal CK2, the pulse audio signal S6, and the pulse signal S9 in order from the top. As shown in S9a to S9g, the pulse signal S9 is generated so that the duty ratio gradually increases with time. S9a is the waveform at time t1 in FIG. 2, and S9g corresponds to the waveform at time t2 in FIG.

The duty ratio adjusting unit 60 starts counting the second clock signal CK2 with reference to the positive edge of the pulse audio signal S6. The pulse width of the pulse signals S9a to S9c is one clock of the second clock signal CK2.
The waveform of the pulse signal S9c is generated by counting the second clock signal CK2 by one pulse for each positive edge of the pulse audio signal S6.
The duty ratio adjusting unit 60 generates the pulse signal S9b and the pulse signal S9a by thinning out the pulse signal S9c at a predetermined ratio (1/2 and 1/4 in FIG. 3).

  Similarly, the pulse widths of the pulse signals S9d to S9g correspond to 2 to 5 pulses of the second clock signal CK2. The control unit 30 instructs the pulse thinning rate and the pulse width by the fourth control signal Sc4.

  Those skilled in the art can use the counter, flip-flop, latch circuit, or digital circuit having a program function to generate the pulse signal S9 shown in FIG. 3 based on the second clock signal CK2 and the pulse audio signal S6. The internal configuration of the duty ratio adjusting unit 60 is not particularly limited.

  Returning to FIG. Over the period of time t1 to t2, the waveform of the pulse signal S9 is changed to S9a to S9g in FIG. 3 by the duty ratio adjusting unit 60. As a result, the analog audio signal S3 gradually increases from the ground level to a voltage corresponding to the predetermined level L1.

  At time t2, the control unit 30 uses the fourth control signal Sc4 to instruct the duty ratio adjustment unit 60 to stop the adjustment control of the pulse width of the pulse audio signal S6. Therefore, after time t2, the duty ratio adjustment unit 60 outputs the pulse audio signal S6 as it is as the pulse signal S9.

  At time t2, the control unit 30 instructs the transition signal generation unit 22 to increase the level of the transition signal S7 by the first control signal Sc1. In response, the transition signal generation unit 22 gradually increases the level of the transition signal S7 toward the midpoint level. After time t2, the pulse width of the pulse signal S9 (S6) increases as the level of the transition signal S7 increases.

  At time t3, the control unit 30 switches the order of the ΔΣ modulator 40 from the second order to the fifth order by the third control signal Sc3. When the transition signal S7 reaches the midpoint level at time t4, a series of activation sequences is completed, and the control unit 30 switches the second control signal Sc2 to the high level. When the second control signal Sc2 becomes high level, the output signal S8 of the selector 24 becomes the interpolated audio signal S5 based on the audio signal S1 from the audio signal generation unit 130. Thereafter, the mute state of the audio signal generation unit 130 is canceled and the sounding state is entered, and the reproduction of the audio signal is started.

  Thus, according to the digital audio processing circuit 100 according to the present embodiment, the level of the analog audio signal S3 can be gradually changed from around 0 V to the intermediate level Vcc / 2, and the occurrence of pop-up noise is suppressed. be able to.

The pulse width of the pulse audio signal S6 generated by the ΔΣ modulator 40 does not become shorter than the pulse width defined by the frequency of the first clock signal CK1. Therefore, when the duty ratio adjusting unit 60 is not provided, the voltage level of the analog audio signal S3 applied to the audio output unit 120 cannot be gradually changed from 0V.
On the other hand, according to the digital audio processing circuit 100 according to the present embodiment, the duty ratio adjusting unit 60 is provided to adjust the duty ratio of the pulse audio signal S6 to a smaller duty ratio. The voltage level of S3 can be changed around 0V, and pop-up noise can be suppressed.

  Furthermore, in the embodiment, the duty ratio adjusting unit 60 changes the duty ratio by thinning out pulses from the pulse audio signal S6 at a predetermined rate. As a result, the duty ratio can be changed more finely than when only the pulse width is changed.

  Although the present invention has been described based on the embodiments, the embodiments merely illustrate the principle and application of the present invention, and the embodiments are intended to include the idea of the present invention defined in the claims. Many modifications and changes in arrangement are possible within the range not leaving.

It is a block diagram which shows the structure of the electronic device carrying the digital audio processing circuit which concerns on embodiment of this invention. 2 is a time chart showing an operation state at the time of starting the digital audio processing circuit of FIG. 1. It is a time chart which shows the adjustment process of the duty ratio of the pulse audio signal by a duty ratio adjustment part.

Explanation of symbols

  10 input interface unit, 20 interpolation filter, 22 transition signal generation unit, 24 selector, 30 control unit, 40 ΔΣ modulator, 50 class D amplifier, 60 duty ratio adjustment unit, 100 digital audio processing circuit, 102 input terminal, 104 output Terminal, 110 low-pass filter, 120 audio output unit, 130 audio signal generation unit, 200 electronic device, Sc1 first control signal, Sc2 second control signal, Sc3 third control signal, Sc4 fourth control signal, S1 input audio signal, S2 output audio signal, S3 analog audio signal, S4 audio signal, S5 interpolated audio signal, S6 pulse audio signal, S7 transition signal, S8 signal, S9 pulse signal, CK1 first clock signal, C 2 the second clock signal.

Claims (12)

A transition signal generator for generating a transition signal at startup;
A ΔΣ modulator that ΔΣ modulates the transition signal;
A duty ratio adjusting unit that adjusts the duty ratio of the bit stream output from the ΔΣ modulator and outputs the same to a subsequent class D amplifier;
An audio processing circuit comprising:   The audio processing circuit according to claim 1, wherein the duty ratio adjustment unit adjusts the duty ratio by changing a pulse width of the bit stream.   3. The duty ratio adjusting unit gradually increases the pulse width of the bit stream from a predetermined minimum value to the pulse width of the input bit stream as time elapses when the circuit is activated. The audio processing circuit according to 1.   The duty ratio adjusting unit receives a second clock signal having a higher frequency than the first clock signal supplied to the ΔΣ modulator, and adjusts the pulse width of the bit stream using the second clock signal. The audio processing circuit according to claim 2 or 3,   The frequency of the first clock signal is m times the sampling frequency of the input audio signal (m is a natural number), and the frequency of the second clock signal is n times the sampling frequency of the input audio signal (n 5 is a natural number satisfying n> m).   4. The audio processing circuit according to claim 1, wherein the duty ratio adjustment unit changes the density by thinning out pulses from the bit stream at a predetermined rate. 5.   4. The selector circuit according to claim 1, further comprising a selector circuit that receives the audio signal to be reproduced and the transition signal generated by the transition signal generation unit, selects one of the signals, and outputs the selected signal to the ΔΣ modulation. The audio processing circuit according to any one of the above. The transition signal generation unit fixes the transition signal to a predetermined level for a predetermined period from the start, and then increases to the midpoint level with time,
The duty ratio adjusting unit is set to an active state in which the transition signal is set to an active state in which the pulse width of the bit stream is changed during the predetermined level, and the bit stream is output as it is during a period in which the level of the transition signal increases. 4. The audio processing circuit according to claim 1, wherein the audio processing circuit is set in an active state.   9. The audio processing circuit according to claim 8, wherein the predetermined level is equal to or lower than a level at which a pulse width of a bit stream generated by a subsequent ΔΣ modulator is minimized.   4. The audio processing circuit according to claim 1, wherein the audio processing circuit is integrated on a single semiconductor substrate. A signal generator for generating a digital audio signal;
The audio processing circuit according to any one of claims 1 to 3, wherein the audio signal is received as an input;
A class D amplifier for amplifying a bit stream from the duty ratio adjustment unit of the audio processing circuit;
A filter for filtering the output signal of the class D amplifier;
An audio output unit provided in a subsequent stage of the filter;
An electronic device comprising: A method of starting an audio processing circuit having a ΔΣ modulator,
Generating a transition signal when the circuit is activated;
ΔΣ-modulating the transition signal;
Adjusting the duty ratio of the bitstream generated by ΔΣ modulation to increase with time;
An activation method comprising:

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