WO2020040068A1

WO2020040068A1 - Sound processing device, sound processing method, and sound processing program

Info

Publication number: WO2020040068A1
Application number: PCT/JP2019/032200
Authority: WO
Inventors: 宜紀田森
Original assignee: ソニー株式会社
Priority date: 2018-08-24
Filing date: 2019-08-16
Publication date: 2020-02-27

Abstract

A sound processing device (100) is provided with: a correction unit (112) that, on the basis of asymmetry of a waveform in one carrier frequency when a predetermined input signal is converted to a pulse width modulation (PWM) signal, corrects the predetermined input signal; and a PWM conversion unit (130) that converts, to a PWM signal, the predetermined input signal having been corrected by the correction unit (112).

Description

Audio processing device, audio processing method, and audio processing program

The present disclosure relates to an audio processing device, an audio processing method, and an audio processing program. More specifically, the present invention relates to a digital audio signal output process.

In recent years, music distribution by high-resolution sound sources, which are audio data with a sound quality exceeding that of music CDs, has been performed. As an example of a high-resolution sound source, a signal recorded in a DSD (Direct Stream Digital) format is used. To reproduce such a digital audio signal, the input signal is converted into a PWM (Pulse Width Modulation) format, and an operating voltage is determined based on the PWM format input signal (hereinafter referred to as a “PWM signal”). A so-called digital amplifier that generates a drive signal by switching is often used.

Here, as a technique relating to a digital amplifier, a technique capable of removing switching distortion and obtaining a reproduced signal as faithful as possible to an original signal is known. For example, in driving a digital amplifier, a waveform of a positive (+) PWM output and a waveform of a negative (−) PWM output corresponding to input data of a PWM signal are linear within one cycle of the carrier frequency. It is considered that a symmetrical waveform is desirable in terms of sound quality.

JP-A-2000-68835

According to the above prior art, it is possible to improve the characteristics of the output audio signal while reducing the circuit scale.

However, in the digital amplifier, the pulse width of the input PWM signal may be limited due to the nature of the amplifier (amplifier circuit). When a signal having a narrow pulse width less than the limit (hereinafter, referred to as “narrow pulse”) is input to a circuit, noise may be generated or a device may be adversely affected. In particular, a high-resolution sound source has a very high sampling frequency, and thus tends to easily generate a narrow pulse. For this reason, in the generation processing of the PWM signal, a signal having a somewhat wide pulse width in which generation of a narrow pulse is suppressed is generated. However, this causes an asymmetric waveform to be generated within one carrier frequency of the PWM signal, or the modulation degree of the PWM signal is reduced. As a result, the output audio may lose its original audio characteristics.

Therefore, the present disclosure proposes an audio processing device, an audio processing method, and an audio processing program that can generate a signal that does not impair the audio characteristics.

In order to solve the above-described problem, an audio processing device according to an embodiment of the present disclosure is based on an asymmetry of a waveform in one carrier frequency when a predetermined input signal is converted into a PWM (Pulse Width Modulation) signal. A correction unit that corrects the predetermined input signal, and a PWM conversion unit that converts the predetermined input signal corrected by the correction unit into a PWM signal.

FIG. 1 is a diagram illustrating a configuration example of an audio processing system according to a first embodiment of the present disclosure. FIG. 3 is a diagram (1) illustrating an example of a waveform of a PWM signal; FIG. 4 is a diagram (2) illustrating an example of a waveform of a PWM signal. FIG. 6 is a diagram (3) illustrating an example of a waveform of a PWM signal. FIG. 4 is a conceptual diagram illustrating an input signal according to the present disclosure by a sine wave. FIG. 3 is a diagram (1) illustrating a correction process according to the present disclosure. FIG. 15 is a diagram (2) illustrating a correction process according to the present disclosure. 5 is a flowchart illustrating a process flow according to the first embodiment of the present disclosure. FIG. 6 is a diagram illustrating a configuration example of a sound processing system according to a second embodiment of the present disclosure. 13 is a flowchart illustrating a flow of a process according to a second embodiment of the present disclosure. FIG. 11 is a diagram for describing an example of a waveform of a PWM signal according to a modified example of the present disclosure. FIG. 3 is a hardware configuration diagram illustrating an example of a computer that realizes functions of a voice processing device.

Hereinafter, embodiments of the present disclosure will be described in detail with reference to the drawings. In the following embodiments, the same portions will be denoted by the same reference numerals, without redundant description.

(1. First Embodiment)
[1-1. Configuration of Speech Processing System According to First Embodiment]
FIG. 1 shows an audio processing system 1 including an audio processing device 100 that executes information processing according to the first embodiment. FIG. 1 is a diagram illustrating a configuration example of the audio processing system 1 according to the first embodiment of the present disclosure. As shown in FIG. 1, the audio processing system 1 includes an audio source 10, an audio output device 20, and an audio processing device 100.

The audio source 10 indicates a medium on which an audio signal processed by the audio processing device 100 is recorded. For example, the audio source 10 holds an audio signal recorded in the DSD format (hereinafter, referred to as “DSD signal”). Then, the audio signal recorded in the audio source 10 is input to the audio processing device 100 via a dedicated playback device or the like.

The DSD signal is a ΔΣ modulated PDM (PulseulDensity Modulation) signal. Although there are a plurality of types of DSD signals depending on the difference in sampling frequency, in the first embodiment, the DSD signal is sampled at 64 times the frequency of 44.1 kHz (hereinafter referred to as “Fs”) which is the sampling frequency of CD. This will be described by taking a DSD signal (referred to as 64 DSD) as an example. In the following description, an audio signal such as a DSD signal may be represented by a combination of a sampling frequency and a bit depth. For example, in the following description, a 64DSD signal may be expressed as [64FS, 1 bit].

The audio output device 20 is a device that outputs a signal output from the audio processing device 100 as sound. For example, the audio output device 20 is a headphone, a speaker, or the like.

The audio processing device 100 is a device that executes information processing according to the present disclosure, and has a function as a so-called digital amplifier that amplifies an audio signal and sends the amplified audio signal to the audio output device 20. Specifically, the audio processing device 100 acquires a DSD signal from the audio source 10 and performs a predetermined correction on the DSD signal. Then, the audio processing device 100 converts the corrected signal into a PWM signal, amplifies the PWM signal with an amplifier unit (the amplification unit 140 illustrated in FIG. 1), and sends the amplified audio signal to the audio output device 20. .

[1-2. Overview of audio processing according to first embodiment]
Subsequently, the outline of the audio processing according to the first embodiment of the present disclosure will be described together with the internal configuration of the audio processing device 100.

As shown in FIG. 1, the audio processing apparatus 100 includes a DSP (Digital Signal Processor) 110, a ΔΣ modulator 120, a PWM conversion unit 130, and an amplification unit 140. Although illustration in FIG. 1 is omitted, the audio processing device 100 operates each processing unit under the control of the control unit and executes information processing according to the present disclosure. The control unit is configured such that a program (for example, an audio processing program according to the present disclosure) stored in the audio processing device 100 is stored in a RAM (Random Access Memory) by a CPU (Central Processing Unit), an MPU (Micro Processing Unit), or the like. And the like as a work area. The control unit may be realized by an integrated circuit such as an ASIC (Application Specific Integrated Circuit) or an FPGA (Field Programmable Gate Array). Note that the audio processing device 100 may include a storage unit (not illustrated in FIG. 1) for storing information to be processed. The storage unit is realized by a semiconductor memory device such as a RAM and a flash memory (Flash @ Memory), or a storage device such as a hard disk and an optical disk.

The DSP 110 performs a predetermined correction process on the input DSD signal. The DSP 110 has a low-pass filter 111 and a correction unit 112.

First, upon acquiring the DSD signal [64 FS, 1 bit] recorded in the audio source 10 (step S11), the audio processing device 100 passes the acquired DSD signal to the low-pass filter 111.

The low-pass filter 111 cuts the high frequency component of the DSD signal, and converts the DSD signal into a PCM signal having a predetermined number of bits as preprocessing of the correction unit 112. In the example of FIG. 1, the low-pass filter 111 converts the DSD signal into a 16-bit PCM signal. Then, the low-pass filter 111 passes the converted PCM signal [64 FS, 16 bits] to the correction unit 112 (Step S12).

The correction unit 112 performs a predetermined correction process on the PCM signal. The details of such correction processing will be described later. The correction unit 112 sends the corrected PCM signal [64 FS, 16 bits] to the ΔΣ modulator 120 (step S13).

The ΔΣ modulator 120 quantizes the PCM signal [64 FS, 16 bit] corrected by the correction unit 112 into a signal [64 FS, 1 bit]. Then, the ΔΣ modulator 120 sends the quantized signal [64 FS, 1 bit] to the PWM conversion unit 130 (Step S14).

The PWM conversion unit 130 converts the signal quantized by the ΔΣ modulator 120 (in other words, the signal corrected by the correction unit 112) [64 FS, 1 bit] into a PWM signal [64 FS, 1 bit]. Then, the PWM conversion unit 130 sends the converted PWM signal [64 FS, 1 bit] to the amplification unit 140 (step S15).

The amplification unit 140 amplifies the PWM signal [64 FS, 1 bit] converted by the PWM conversion unit 130 and generates an output signal that is a signal output as sound. The amplification unit 140 is a processing unit corresponding to an amplification circuit of a so-called digital amplifier. That is, the amplifying unit 140 receives a PWM signal whose cycle is constant and whose duty cycle of the pulse width changes according to the level (magnitude) of the input signal, and performs switching control based on the PWM signal, Generate an output signal. For example, in the present disclosure, the amplification unit 140 generates an output signal based on a PWM signal having two types of pulse widths illustrated in FIG. Then, the amplifier 140 sends the generated output signal to the audio output device 20 (Step S16).

The audio output device 20 outputs audio corresponding to the DSD signal stored in the audio source 10 based on the output signal sent from the amplifier 140.

As described above, the sound processing apparatus 100 corrects a predetermined input signal by the correction unit 112, and converts the corrected signal into a PWM signal. Details of this processing will be described with reference to FIG.

FIG. 2A is a diagram (1) illustrating an example of a waveform of a PWM signal. A waveform 30 shown in FIG. 2A shows an example of a waveform when a predetermined DSD signal [64 FS, 1 bit] is not converted to a PWM signal as a PDM signal. The optimum shape of the waveform of the PWM signal differs depending on the driver driving method. However, in the first embodiment, it is assumed that the driver driving method employs single-ended driving.

The DSD signal outputs either “0” or “1” every 64 Fs which is the sampling frequency. The output may be read as a pulse. Since the DSD signal is a signal modulated by pulse density modulation, the sound corresponding to the DSD signal is determined by the density of the output pulse.

Here, for the sake of explanation, when the DSD signal is converted into a PWM signal, a signal corresponding to “1” in the DSD signal is represented by PMW (+), and a signal corresponding to “0” in the DSD signal is represented by PMW (−). Called. In the example of FIG. 2A, the waveform 38 indicated by "1" corresponds to PMW (+), and the waveform 36 indicated by "0" corresponds to PMW (-).

Also, the shape of the waveform of the PWM signal is determined by the resolution, and this resolution is referred to as “slot” in this specification. The resolution is uniquely determined by the master clock that controls the shape of the PWM signal and the carrier frequency of the PWM signal, that is, the sampling frequency of the DSD signal. In the first embodiment, it is assumed that the master clock is “1024Fs”. In this case, the resolution is 1024 FS / 64 Fs = 16. This indicates that 16 slots are included in one carrier frequency. In the example of FIG. 2A, width 32 corresponds to a sampling frequency (carrier frequency), and width 34 corresponds to a slot.

As described above, the optimum waveform of the PWM signal in the audio characteristics is that the waveforms of PWM (+) and PWM (-) are each line-symmetric with respect to the center within one carrier frequency.

Here, referring to the waveform 30 in FIG. 2A, PWM (+) occupies 16 slots and PWM (-) occupies 0 slot. Although this is line-symmetric, no pulse change is observed within one carrier, and the period at which the edge of the pulse occurs cannot be fixed. Therefore, from the viewpoint of audio performance, as a drive signal for the amplifier corresponding to the DSD signal, Actually, the waveform 30 does not tend to be generated.

Therefore, the audio processing device 100 generates at least the PWM signal shown in FIG. 2B. FIG. 2B is a diagram (2) illustrating an example of a waveform of a PWM signal. A waveform 40 shown in FIG. 2B is a waveform when a predetermined DSD signal [64 FS, 1 bit] is converted into a PWM signal, and is a waveform in which PWM (-) has a minimum width (two slots). The width 42 and the width 44 are the same as the width 32 and the width 34 shown in FIG. 2A, respectively.

If PWM (+) maintains a line-symmetric shape when PWM (-) has a width of 2 slots, PWM (+) will have a width of 14 slots. As shown in FIG. 2B, waveform 46 shows a shape in which PWM (-) has a pulse width of 2 slots. The waveform 48 indicates a shape in which PWM (+) has a pulse width of 14 slots.

Since the pulse width of the waveform 46 is equivalent to two slots, the pulse width is expressed as (1/1024 Fs) × 2 ≒ 44 nanoseconds in terms of time. Here, as described above, the amplifier 140 of the digital amplifier has a limit on the pulse width that can be input, and the limit is assumed to be 50 nanoseconds. In this case, since a pulse shorter than 50 nanoseconds cannot be input, a desired DSD signal cannot be driven unless the specifications of the amplifier 140 are changed or replaced with a different circuit. That is, the waveform 40 shown in FIG. 2B shows a waveform when a narrow pulse is generated.

Therefore, when trying to process the waveform 40 shown in FIG. 2B, the sound processing device 100 executes a process for avoiding a narrow pulse. For example, the waveform 40 shown in FIG. 2B is converted into the waveform shown in FIG. 2C. FIG. 2C is a diagram (3) illustrating an example of a waveform of a PWM signal. A waveform 50 shown in FIG. 2C is a waveform when a predetermined DSD signal [64 FS, 1 bit] is converted into a PWM signal, and the width of PWM (-) when avoiding a narrow pulse is the minimum width (4 slots). This is the waveform. The width 52 and the width 54 are the same as the width 32 and the width 34 shown in FIG. 2A, respectively.

If PWM (+) maintains a line-symmetric shape when PWM (−) has a width of 4 slots, PWM (+) has a width of 12 slots by nature. However, it is known that when the PWM (+) slot width is reduced in this manner, the modulation degree of a signal is reduced. The degree of modulation is obtained by subtracting the resolution of one of the PWM signals from the resolution of one of the PWM signals, using the overall resolution at one carrier frequency as the denominator. For example, if PWM (+) has 16 slots and M (-) has 0 slots, (16/16)-(0/16) = 1. As described above, if PWM (-) has 4 slots and PWM (+) has 12 slots, the modulation degree is (12/16)-(4/16) = 0.5.

2A to 2C show examples in which the master clock is 1024 Fs, but consider an example in which the master clock is 512 Fs. In this case, since the resolution of one carrier frequency is eight slots, the PWM (-) is set to four slots as described above (in this example, the limitation of the pulse width input to the amplifier 140 is, for example, 100 nanoseconds or the like). ), And PWM (+) also has 4 slots if it is desired to maintain a line-symmetric shape. In this case, the waveforms of PWM (-) and PWM (+) have the same shape, and the respective signals are in a state of 50% duty within one carrier frequency, so that a so-called mute state is obtained. This state indicates that conversion to a PWM signal is impossible.

Therefore, the sound processing device 100 according to the present disclosure generates the waveform 50 illustrated in FIG. 2C. In the waveform 50, the waveform 56 indicating PWM (-) has a pulse width of 4 slots to avoid a narrow pulse. On the other hand, the waveform 58 indicating PWM (+) maintains the 14-slot pulse width, similarly to the waveform 48 of FIG. 2B, although the PWM (-) has increased the pulse width to avoid narrow pulses. That is, the waveform 50 has a waveform in which PWM (−) and PWM (+) are asymmetric.

As described above, if PWM (−) and PWM (+) are asymmetric within one carrier frequency, audio characteristics may be degraded. This will be described with reference to FIG.

FIG. 3 is a conceptual diagram showing a sine wave of an input signal according to the present disclosure. FIG. 3 conceptually shows a sine wave in the case where an input signal (PWM signal or DSD signal) is converted into an audio waveform for explanation. Note that the audio processing apparatus 100 can obtain a sine wave as shown in FIG. 3 by passing a PWM signal, a DSD signal, or the like (pulse wave) through a predetermined low-pass filter, for example.

波形 The waveform 62 shown in FIG. 3 indicates a case where PWM (−) and PWM (+) are symmetric within one carrier frequency, that is, a sine wave which is an ideal voice waveform. As shown in FIG. 3, the waveform 62 has the same absolute value of the upper and lower amplitudes (“0.5” in the example of FIG. 3) because the PWM (−) and the PWM (+) are symmetric. . Specifically, the waveform 62 is an ideal waveform in which PWM (−) and PWM (+) are symmetrical within one carrier frequency, as shown in FIGS. 2A and 2B, as a sine wave. . For example, the waveform 62 shows the original DSD signal held by the audio source 10 as a sine wave.

On the other hand, a waveform 61 shown in FIG. 3 shows a sine wave when PWM (−) and PWM (+) are asymmetric within one carrier frequency, that is, when the audio characteristics are degraded. As shown in FIG. 3, the waveform 61 has different absolute values of the upper and lower amplitudes because the PWM (−) and the PWM (+) are asymmetric. Specifically, in the waveform 61, the level on the plus side is “0.5”, while the level on the minus side is “− (0.5 × N (N is a predetermined level determined by the degree of asymmetry. Number)) ". As described above, when a sine wave having different upper and lower amplitudes is output as a sound, a phenomenon such as reproduction of unnatural overtones or distortion occurs, and audio characteristics are deteriorated. For example, the waveform 61 indicates a PWM signal as a sine wave when the correction processing by the DSP 110 according to the present disclosure has not been performed.

を Referring to FIG. 3, the waveform 61 has a waveform 62 whose amplitude on the minus side is reduced from “−0.5” to “− (0.5 × N)”. This is because an asymmetric PWM signal is generated in order to suppress the generation of a narrow pulse as described above.

Here, if the negative side amplitude of the sine wave increases due to the generation of an asymmetric waveform in the PWM conversion unit 130, some correction is performed on the input signal before input to the PWM conversion unit 130, It is possible to finally adjust the upper and lower amplitudes to have the same value. More specifically, the audio processing apparatus 100 obtains an input signal that takes into account (corrected) the rise from the beginning before inputting the original DSD signal to the PWM conversion unit 130, so that the PWM processing unit 130 After the conversion, a PWM signal corresponding to a sine wave having the same upper and lower amplitudes can be obtained.

波形 The waveform 63 in FIG. 3 is a sine wave corresponding to the input signal after the above-described correction processing. As shown in FIG. 3, the waveform 63 is a sine wave whose level on the plus side is “0.5” and whose level on the minus side is “− (0.5 ÷ N)”. That is, the waveform 63 represents a signal input to the PWM conversion unit 130 as a sine wave after the correction processing by the DSP 110. Specifically, such a signal is a signal transmitted in step S13 shown in FIG. 1 represented by a sine wave.

As described with reference to FIG. 3, when the correction processing according to the present disclosure is not performed, when the PWM signal in which the generation of the narrow pulse is suppressed is converted, the pulse width on the PWM (−) side increases. The amplitude on the minus side indicated by the symbol decreases, and the amplitude of the sine wave becomes asymmetric. As a result, audio characteristics are degraded, such as distortion. Therefore, the audio processing device 100 according to the present disclosure performs correction to an input signal corresponding to the waveform 63 shown in FIG. 3 in advance to prevent a decrease in audio characteristics in a finally output signal. Can be.

Hereinafter, the correction process performed by the audio processing device 100 will be specifically described with reference to FIGS. 4 and 5. FIG. 4 is a diagram (1) illustrating a correction process according to the present disclosure. FIG. 4 shows the waveforms of the PWM signals shown in FIGS. 2A and 2B and the sine waves corresponding to the PWM signals, respectively.

4) The waveform 30 shown in FIG. 4 corresponds to the waveform 30 shown in FIG. 2A. The sine wave 65 is a sine wave of the PWM signal represented by the waveform 30. In the example shown in FIG. 4, a waveform 30 indicates a PWM signal in the case where PWM (+) and PWM (−) can have the maximum pulse width. That is, when expressing the waveform 30 as the sine wave 65, the absolute value of the amplitude becomes the maximum value (in the example of FIG. 4, it is assumed to be "0.5").

に As described above, the waveform 30 tends to be not actually used as a PWM signal because the period at which a pulse edge occurs within one carrier frequency cannot be fixed. Therefore, the audio processing device 100 adds two slots having the minimum pulse width to the PWM (-) (step S21). Thereby, the audio processing device 100 obtains the waveform 40 shown in FIG. 4 as a PWM signal. Waveform 40 corresponds to waveform 40 shown in FIG. 2B. The sine wave 66 is a sine wave of the PWM signal indicated by the waveform 40.

では In the example shown in FIG. 4, in the waveform 40, the pulse width of PWM (+) is “14/16” and the pulse width of PWM (−) is “2/16”. In this case, the amplitude of the sine wave 66 is obtained by multiplying the modulation degree by the maximum value of the amplitude. Since the modulation degree of the waveform 40 is “(14/16) − (2/16) = 0.75”, the absolute value of the amplitude of the sine wave 66 is “0.5 × 0.75 = 0.375”. Becomes In the waveform 40, since the PWM (+) and the PWM (-) are symmetrical, the amplitude of the sine wave 66 has the same absolute value of "0.375" on the plus side and the minus side.

Here, assuming that the PWM (-) shown in the waveform 40 is a narrow pulse, the audio processing device 100 generates a waveform that is expanded so that the PWM (-) does not become a narrow pulse (step S22). Thereby, the audio processing device 100 obtains the waveform 50 shown in FIG. 4 as a PWM signal. Waveform 50 corresponds to waveform 50 shown in FIG. 2C. The sine wave 67 is a sine wave of the PWM signal indicated by the waveform 50.

In the example shown in FIG. 4, the waveform 50 has a PWM (+) pulse width of “14/16” and a PWM (−) pulse width of “4/16”. In the waveform 50, since the PWM (+) and the PWM (-) are asymmetric, the amplitude of the corresponding sine wave 67 has different absolute values on the plus side and the minus side.

In this case, the plus side amplitude of the sine wave 67 is obtained from the modulation factor on the assumption that a waveform symmetrical to the PWM (+) side pulse width is obtained in the waveform 50. Specifically, the pulse width of the PWM (+) of the waveform 50 is “14/16”, and considering the pulse width of the PWM (−) that is symmetrical to this, the pulse width of the temporary PWM (−) is "2/16". For this reason, the plus side amplitude of the sine wave 67 is obtained by multiplying the temporary modulation factor “(14/16) − (2/16) = 0.75” by the maximum value “0.5.” 5 × 0.75 = 0.375 ”.

On the other hand, the negative amplitude of the sine wave 67 can be obtained from the modulation factor on the assumption that a waveform symmetrical to the pulse width on the PWM (-) side is obtained in the waveform 50. Specifically, the pulse width of the PWM (−) of the waveform 50 is “4/16”, and considering the pulse width of the PWM (+) which is symmetrical to this, the pulse width of the temporary PWM (+) is It becomes “12/16”. For this reason, the amplitude on the minus side of the sine wave 67 is obtained by multiplying the temporary modulation factor “(12/16) − (4/16) = 0.5” by the maximum value “0.5.” 5 × 0.5 = 0.25 ”.

From the above, the sine wave 67 is an asymmetric waveform having a positive amplitude “0.375” and a negative amplitude “−0.25”.

補正 Based on the description of FIG. 4, a correction process performed by the audio processing device 100 will be described with reference to FIG. FIG. 5 is a diagram (2) illustrating a correction process according to the present disclosure.

波形 The waveform 30 shown in FIG. 5 corresponds to the waveform 30 shown in FIG. Further, the sine wave 65 shown in FIG. 5 corresponds to the sine wave 65 shown in FIG.

When the DSP 110 receives the DSD signal corresponding to the waveform 30 by the DSP 110 (step S11 shown in FIG. 1), the audio processing apparatus 100 converts the DSD signal into a 16-bit sine wave by passing through the low-pass filter 111. As a result, the amplitude can be corrected as described later.

{Circle around (4)} Then, the sound processing device 100 passes the waveform 30 passed through the low-pass filter 111 (more precisely, the input signal indicated by the waveform 30) to the correction unit 112 (step S31).

The correction unit 112 corrects the input signal received in step S31 in order to prevent the amplitude of the sine wave converted into the PWM signal by the PWM conversion unit 130 from becoming asymmetric. For example, the correction unit 112 corrects the input signal based on the asymmetry of the waveform within one carrier frequency when the input signal is converted into a PWM signal.

{Specifically, the correction unit 112 corrects a value corresponding to the amplitude of the input signal based on the asymmetry of the waveform within one carrier frequency when the input signal is converted into a PWM signal. More specifically, the correction unit 112 corrects the input signal based on the modulation factor when it is assumed that the input signal is converted into a PWM signal so that the waveform within one carrier frequency is symmetric.

As described with reference to FIG. 4, in the asymmetrical waveform 50 having the minimum pulse width in which the generation of the narrow pulse is suppressed, the sine wave 67 has “0.375” on the plus side and “−0.0. 25 ". Of these, the plus side is not corrected because it is the maximum value of PWM (+) that can be assumed when PWM (−) is the minimum (narrow pulse) width. On the other hand, the minus side is obtained by expanding PWM (-) from the minimum width (narrow pulse) to 4 slots, and there is room for correction. Therefore, the audio processing device 100 corrects the value of the amplitude corresponding to the minus side.

As described with reference to FIG. 4, the amplitude on the minus side is obtained from the modulation factor on the assumption that a waveform symmetrical to the pulse width on the PWM (−) side is obtained in the waveform 50. That is, the modulation degree is “(12/16) − (4/16) = 0.5”. Further, as described above, the absolute value of the amplitude on the positive side of the sine wave after conversion into the PWM signal is “0.375”.

For this reason, the correction unit 112 calculates the negative-side amplitude of the sine wave 71 corresponding to the signal subjected to the correction processing by calculating backward from “−0.375” which is the target value of the absolute value of the positive-side amplitude. The input signal is corrected so that “−0.375 ÷ 0.5 = −0.75”. Specifically, the correction unit 112 corrects the bit string of the input signal of [64 FS, 16 bits] through the low-pass filter 111. For example, the correction unit 112 corrects the bit sequence of the input signal by multiplying the bit sequence of the input signal by a predetermined numerical value using a predetermined multiplier. Thereby, the correction unit 112 obtains a signal corresponding to the sine wave 71 shown in FIG.

The correction unit 112 sends the corrected input signal to the ΔΣ modulator 120 and quantizes the input signal, and then sends the input signal to the PWM conversion unit 130 (step S32).

The PWM conversion unit 130 generates a waveform 51 in which PWM (+) and PWM (−) are asymmetric in order to avoid a narrow pulse. However, since the waveform 51 has been subjected to the correction processing by the correction unit 112, the sine wave 72 corresponding to the waveform 51 has the same absolute value on the plus side and the minus side. Specifically, the sine wave 72 is shown as a sine wave having the same absolute value “0.375” in amplitude on the plus side and the minus side.

As a result, the audio processing apparatus 100 has a symmetrical amplitude in order to avoid a narrow pulse even if the waveform within one carrier frequency of the PWM signal is asymmetric between PWM (+) and PWM (-). Sine wave can be obtained.

As described above, the correction unit 112 corresponds to the amplitude of the input signal based on the modulation factor when it is assumed that the input signal is converted into the PWM signal so that the waveform within one carrier frequency is symmetric. Increase the value. For example, the correction unit 112 corrects the input signal so as to obtain the sine wave 75 in which the absolute value of the negative amplitude of the sine wave 65 shown in FIG. 5 is increased from “0.5” to “0.75”. I do. Thus, the audio processing device 100 can obtain an audio output signal whose audio characteristics are not deteriorated even from a PWM signal that is asymmetric to avoid a narrow pulse.

As described above, the DSD signal is sampled at a very high frequency, and since there are various types of sampling frequencies, it is difficult for a general digital amplifier to avoid narrow pulses or reproduce a sound source. Or may be. However, according to the audio processing device 100 according to the present disclosure, it is possible to input a variety of DSD signals (sound sources) because the signal can be converted into a PWM signal that does not impair the audio characteristics while avoiding narrow pulses.

{Also, as described above, the audio processing according to the present disclosure is realized by a configuration in which the DSP 110 and the ΔΣ modulator 120 are installed before the PWM conversion unit 130, and thus does not depend on the performance of the amplification unit 140. This means that a digital amplifier can be manufactured without depending on the performance (characteristics) of the amplifier 140. That is, according to the audio processing according to the present disclosure, the options of the amplification unit 140 can be significantly expanded.

Note that the audio processing apparatus 100 receives information (a pulse width that can be input, etc.) relating to the amplifying unit 140 and a value of the master clock in advance, so that the temporary processing when the narrow pulse is avoided as described above. The degree of modulation can be determined. The sound processing device 100 performs the above-described correction processing based on the provisional modulation degree. Note that the audio processing device 100 may acquire the characteristics and the like of the amplification unit 140 in real time, dynamically calculate the temporary modulation degree, and execute the above-described correction processing. Details of such processing will be described in a second embodiment.

[1-3. Information processing procedure according to first embodiment]
Next, an information processing procedure according to the first embodiment will be described with reference to FIG. FIG. 6 is a flowchart illustrating a process flow according to the first embodiment of the present disclosure. Specifically, FIG. 6 illustrates a flow of a process executed by the audio processing device 100 according to the first embodiment.

(6) As shown in FIG. 6, the audio processing device 100 determines whether an input of a DSD signal has been received (step S101). When the input of the DSD signal is not received (Step S101; No), the audio processing device 100 waits until the input of the DSD signal is received.

On the other hand, when the input of the DSD signal is received (Step S101; Yes), the audio processing device 100 converts the received DSD signal into a PCM signal (Step S102).

Subsequently, the audio processing device 100 corrects the PCM signal based on the asymmetry of the waveform when the DSD signal received in step S101 is converted into a PWM signal (step S103). Specifically, the audio processing device 100 calculates a temporary modulation factor based on the asymmetry of the waveform, and corrects the bit string of the PCM signal based on the calculated modulation factor.

The audio processing device 100 inputs the corrected PCM signal to the ΔΣ modulator 120, and quantizes it (step S104). Subsequently, the audio processing device 100 inputs the quantized input signal to the PWM conversion unit 130 and converts it into a PWM signal (step S105).

Thereafter, in the audio processing device 100, the amplifier 140 generates an output signal from the PWM signal (step S106). As described above, the audio processing apparatus 100 extracts the audio signal from the input signal that has been subjected to the correction processing based on the asymmetry of the waveform in advance, thereby enabling audio output without deteriorating the audio characteristics.

[1-4. Modification Example of First Embodiment]
In the first embodiment, an example has been described in which the audio processing device 100 receives a DSD format signal from the audio source 10 and performs a correction process on the received DSD signal. However, the signal corrected by the audio processing device 100 is not limited to the DSD format, and may be any format signal as long as the signal is quantized by pulse density modulation.

In the first embodiment, the example in which the audio processing apparatus 100 converts the DSD signal into the 16-bit PCM signal in the DSP 110 has been described. However, the audio processing apparatus 100 may convert the DSD signal into a 16-bit PCM signal. That is, the audio processing device 100 may convert the DSD signal into a PCM signal other than 16 bits as long as a signal of a bit sequence that can express the correction processing by the correction unit 112 is obtained.

(2. Second Embodiment)
[2-1. Configuration of audio processing system according to second embodiment]
Next, a second embodiment will be described with reference to FIGS. 7 and 8. FIG. 7 is a diagram illustrating a configuration example of the audio processing system 2 according to the second embodiment of the present disclosure. As illustrated in FIG. 7, the audio processing device 100A according to the second embodiment further includes an analysis unit 150, a switch 160, and a switch 170 as compared with the first embodiment. In the following, description of the same configuration as that according to the first embodiment will be omitted.

In the first embodiment, the audio processing apparatus 100 generates an asymmetric waveform in advance based on the characteristics of the amplifying unit 140 (eg, a pulse width that allows input), the sampling frequency of the DSD signal, and the value of the master clock. An example has been shown in which the degree of modulation is calculated and the DSP 110 performs correction based on the calculated degree of modulation.

The audio processing device 100A according to the second embodiment obtains variables such as the characteristics of the above-described amplifying unit 140 and analyzes those values to enable dynamic correction processing. Specifically, the analysis unit 150 according to the audio processing device 100A acquires various variables in the process of information processing performed by the audio processing system 2, analyzes the acquired values, and performs dynamic correction processing. Execute

For example, the analysis unit 150 analyzes the driving ability of the amplification unit 140 that drives the PWM signal converted by the PWM conversion unit 130. Specifically, the analysis unit 150 analyzes the pulse width of the PWM signal allowed by the amplification unit 140 as the driving capability.

For example, in the process of the information processing described in the first embodiment, the analysis unit 150 determines the waveform of the PWM signal when the narrow pulse is avoided and the PWM signal is symmetric, and the value of the modulation factor or the like. It is acquired (step S51). Also, in the process of the information processing shown in the first embodiment, the analysis unit 150 determines the waveform of the PWM signal when the PWM signal is not asymmetric and the PWM signal is asymmetric, To get. Further, the analysis unit 150 acquires a set value of the master clock in the audio processing, and the like.

Further, the analysis unit 150 acquires the audio output signal output from the amplification unit 140, and analyzes whether the acquired waveform is a symmetrical waveform, whether or not distortion has occurred (step S1). S52). In addition, the analysis unit 150 acquires information such as impedance in the audio output device 20 (Step S53). In addition, the analysis unit 150 acquires the sampling frequency of the DSD signal input to the audio processing device 100A, the type of the DSD file, and the like from the audio source 10 (step S54).

{Then, the analysis unit 150 dynamically determines whether or not the DSP 110 should correct the input signal based on the acquired information. In addition, the analysis unit 150 dynamically calculates a correction value for correction.

For example, when the pulse width allowed by the amplification unit 140 can be obtained, the analysis unit 150 avoids the narrow pulse from the setting value of the master clock and the sampling frequency of the DSD signal, as in the first embodiment. The degree of modulation in the case can be calculated. Therefore, according to the audio processing device 100A, it is possible to perform correction with an appropriate value without setting the modulation degree and the like to the DSP 110 in advance.

According to the analysis unit 150, since the impedance value of the audio output device 20 can be acquired, a process of adjusting the correction value according to the audio output device 20 can be performed. For example, as the value of the impedance decreases, the value of the current flowing increases, so that the load applied to the amplifier 140 increases. Specifically, the load of the amplification process of the amplification unit 140 increases. In such a case, the analysis unit 150 increases (thickers) the minimum pulse width input to the amplification unit 140 from the current set value. This eliminates the need for the amplification section 140 to perform amplification with a narrow pulse width, and thus reduces the load of amplification. Then, when performing an adjustment to increase the minimum pulse width, the analysis unit 150 newly calculates a correction value corresponding to the changed pulse width.

For example, in the first embodiment, the minimum pulse width when avoiding a narrow pulse is “4 slots”. However, if the width is changed to “6 slots”, the analysis unit 150 A correction value (a value determined based on the modulation factor) that changes accordingly is newly calculated. Then, the analysis unit 150 sets the newly calculated correction value in the DSP 110 (correction unit 112).

That is, the correction unit 112 calculates a modulation factor based on the pulse width analyzed by the analysis unit 150, assuming that a predetermined input signal is converted into a PWM signal such that a waveform within one carrier frequency is symmetric. Then, the input signal is corrected based on the calculated degree of modulation. As described above, according to the analysis unit 150, it is possible to analyze the structure of the entire circuit in the audio processing system 2 and dynamically determine an optimal correction value.

In addition, in the information processing according to the present disclosure, the analysis unit 150 may determine that the correction processing is to be skipped. For example, it is assumed that the analysis unit 150 analyzes the sampling frequency of the DSD signal acquired from the audio source 10, the set value of the master clock, and the like, and determines that a narrow pulse cannot be generated in the PWM conversion. For example, it is assumed that the analysis unit 150 determines that the minimum pulse width generated in processing a certain DSD signal is always larger than the pulse width permitted by the amplification unit 140. In this case, since the correction process by the correction unit 112 is not required, the process of converting the DSD signal into a PCM signal or the like and the process of re-quantization are wasted.

If the pulse width satisfies the predetermined condition as described above, the analysis unit 150 may cause the correction process to be skipped by switching the switches 160 and 170 (step S55). Specifically, when the analysis unit 150 determines that the minimum pulse width satisfies the predetermined condition, the analysis unit 150 switches the switch 160 and sends the DSD signal received from the audio source 10 directly to the PWM conversion unit 130 (Step S56). Further, analysis section 150 switches switch 170 and sends the signal output from PWM conversion section 130 to amplification section 140. If the correction processing is not to be skipped, the analysis unit 150 switches the switch 160 to the DSP 110 (step S57).

That is, when the pulse width analyzed by the analysis unit 150 satisfies the predetermined condition, the correction unit 112 does not correct the predetermined input signal. Further, the PWM conversion unit 130 converts a predetermined input signal before correction by the correction unit 112 (that is, the original DSD signal recorded on the audio source 10) into a PWM signal.

Thus, when the correction processing is not performed, the audio processing apparatus 100A can perform the processing without the intervention of the DSP 110 or the like, so that the processing speed can be increased and the processing load can be reduced.

[2-2. Procedure of information processing according to second embodiment]
Next, an information processing procedure according to the second embodiment will be described with reference to FIG. FIG. 8 is a flowchart illustrating a flow of a process according to the second embodiment of the present disclosure.

As shown in FIG. 8, the audio processing device 100A analyzes information related to the narrow pulse based on the characteristics of the amplifying unit 140, the set value of the master clock, the sampling frequency of the DSD signal, and the like (step S201).

Then, the audio processing device 100A determines whether the input of the DSD signal has been received (step S202). If the input of the DSD signal has not been received (Step S202; No), the audio processing device 100A waits until the input of the DSD signal is received.

On the other hand, when the input of the DSD signal is received (Step S202; Yes), the audio processing device 100A determines whether the received DSD signal needs to be corrected (Step S203).

If it is determined that the DSD signal needs to be corrected (step S203; Yes), the audio processing device 100A converts the DSD signal into a PCM signal (step S204). Then, the audio processing device 100A generates the PCM signal based on the asymmetry of the waveform when the DSD signal is converted into the PWM signal (in the second embodiment, information (modulation degree) analyzed by the analysis unit 150). Is corrected (step S205).

The audio processing device 100A inputs the corrected PCM signal to the ΔΣ modulator 120 and quantizes it (step S206). Subsequently, the audio processing device 100A inputs the quantized input signal to the PWM conversion unit 130, and converts the input signal into a PWM signal (step S207). Note that when the audio processing device 100 determines that the correction process on the DSD signal is not necessary (Step S203; No), the process from Step S204 to Step S206 is skipped, and the process of Step S207 is executed.

Then, in the audio processing device 100A, the amplifier 140 generates an output signal from the PWM signal (step S207). As described above, since the audio processing device 100A can dynamically perform the correction process by analyzing the variables related to the audio processing system 2, the information processing of the present disclosure is applied to various types of DSD signals. Or the design of the circuit of the amplifier 140 can be flexibly changed.

(3. Other embodiments)
The processing according to each of the above-described embodiments may be performed in various different forms other than the above-described embodiments. In the following, unless otherwise specified, the content including both the audio processing device 100 and the audio processing device 100A is simply described as the audio processing device 100.

In each of the above embodiments, an example is described in which the audio processing device 100 drives the PWM signal by a single-end drive method, but the drive may be a balance drive method. This will be described with reference to FIG. FIG. 9 is a diagram illustrating an example of a waveform of a PWM signal according to a modified example of the present disclosure.

波形 A waveform 81 shown in FIG. 9 shows an example of a PWM signal waveform in the balance driving method. In the case of the single drive system, as shown in FIG. 2A and the like, the PWM signal has symmetry at the center within one carrier frequency. On the other hand, in the balance driving method, as shown by a waveform 81, a waveform obtained by differentially combining a positive signal and a negative signal has symmetry. In the example of FIG. 9, a positive signal of PWM (+) (“1” shown in FIG. 9) among the PWM signals is described as “PWM (+ P)”, and a negative signal is described as “PWM (+ N)”. I do. Further, among the PWM signals, a positive signal of PWM (−) (“0” shown in FIG. 9) is expressed as “PWM (−P)”, and a negative signal is expressed as “PWM (−N)”. Note that the waveform 81 indicates the minimum pulse width in PWM (-P) and PWM (+ N), and corresponds to the narrow pulse waveform shown in FIG. 2B.

場合 When the pulse widths of PWM (−P) and PWM (+ N) are expanded to avoid the narrow pulse of the waveform 81 (step S60), the audio processing device 100 obtains the waveform 82 shown in FIG. A waveform 82 shown in FIG. 9 is an example of a waveform obtained by expanding the pulse widths of PWM (−P) and PWM (+ N) in order to avoid a narrow pulse. As shown in FIG. 9, in the waveform 82, the PWM (−P) and the PWM (−N) in one carrier are expanded by expanding the pulse widths of the PWM (−P) and the PWM (+ N) with respect to the waveform 81. ) And the waveform of the differential combination of PWM (+ P) and PWM (+ N) in one carrier are asymmetric. As shown in FIG. 9, although the waveform of the PWM signal is different in the case of the balanced drive system from the case of the single-end drive system, the audio processing apparatus 100 performs the final waveform ( For example, it is possible to perform correction so that the amplitude of a sound output wave represented by a sine wave is symmetric. That is, the audio processing device 100 can realize audio processing that does not impair the audio characteristics, regardless of the driving method.

In each of the above embodiments, the configuration in which the audio processing device 100 includes the amplification unit 140 has been described. However, the amplification unit 140 may be configured as a single digital amplifier, or may be configured to be included in the audio output device 20.

Further, among the processes described in the above embodiments, all or a part of the processes described as being performed automatically may be manually performed, or the processes described as being performed manually may be performed. Can be automatically or entirely performed by a known method. In addition, the processing procedures, specific names, and information including various data and parameters shown in the above documents and drawings can be arbitrarily changed unless otherwise specified. For example, the various information shown in each drawing is not limited to the information shown.

The components of each device shown in the drawings are functionally conceptual, and do not necessarily need to be physically configured as shown in the drawings. That is, the specific form of distribution / integration of each device is not limited to the one shown in the figure, and all or a part thereof may be functionally or physically distributed / arbitrarily divided into arbitrary units according to various loads and usage conditions. Can be integrated and configured.

The embodiments and the modifications described above can be combined as appropriate within a range that does not contradict processing contents.

効果 In addition, the effects described in this specification are merely examples and are not limited, and other effects may be provided.

(4. Hardware configuration)
The information devices such as the voice processing device 100 according to each embodiment described above are realized by, for example, a computer 1000 having a configuration as shown in FIG. Hereinafter, the audio processing device 100 according to the first embodiment will be described as an example. FIG. 10 is a hardware configuration diagram illustrating an example of a computer 1000 that implements the functions of the audio processing device 100. The computer 1000 has a CPU 1100, a RAM 1200, a ROM (Read Only Memory) 1300, a HDD (Hard Disk Drive) 1400, a communication interface 1500, and an input / output interface 1600. Each unit of the computer 1000 is connected by a bus 1050.

The CPU 1100 operates based on a program stored in the ROM 1300 or the HDD 1400, and controls each unit. For example, the CPU 1100 loads a program stored in the ROM 1300 or the HDD 1400 into the RAM 1200, and executes processing corresponding to various programs.

The ROM 1300 stores a boot program such as a BIOS (Basic Input Output System) executed by the CPU 1100 when the computer 1000 starts up, a program that depends on the hardware of the computer 1000, and the like.

The HDD 1400 is a computer-readable recording medium for non-temporarily recording a program executed by the CPU 1100 and data used by the program. Specifically, HDD 1400 is a recording medium that records an audio processing program according to the present disclosure, which is an example of program data 1450.

The communication interface 1500 is an interface for connecting the computer 1000 to an external network 1550 (for example, the Internet). For example, the CPU 1100 receives data from another device via the communication interface 1500 or transmits data generated by the CPU 1100 to another device.

The input / output interface 1600 is an interface for connecting the input / output device 1650 and the computer 1000. For example, the CPU 1100 receives data from an input device such as a keyboard and a mouse via the input / output interface 1600. In addition, the CPU 1100 transmits data to an output device such as a display, a speaker, or a printer via the input / output interface 1600. Further, the input / output interface 1600 may function as a media interface that reads a program or the like recorded on a predetermined recording medium (media). The medium is, for example, an optical recording medium such as a DVD (Digital Versatile Disc), a PD (Phase changeable rewritable Disk), a magneto-optical recording medium such as an MO (Magneto-Optical disk), a tape medium, a magnetic recording medium, or a semiconductor memory. It is.

For example, when the computer 1000 functions as the audio processing device 100 according to the first embodiment, the CPU 1100 of the computer 1000 implements the functions of the DSP 110 and the like by executing the audio processing program loaded on the RAM 1200. Further, the HDD 1400 stores an audio processing program according to the present disclosure and data such as a DSD signal to be subjected to audio processing. Note that the CPU 1100 reads and executes the program data 1450 from the HDD 1400. However, as another example, the CPU 1100 may acquire these programs from another device via the external network 1550.

Note that the present technology can also have the following configurations.
(1)
A correction unit that corrects the predetermined input signal based on the asymmetry of the waveform within one carrier frequency when the predetermined input signal is converted into a PWM (Pulse Width Modulation) signal;
A PWM conversion unit that converts a predetermined input signal corrected by the correction unit into a PWM signal.
(2)
The correction unit,
The audio processing device according to (1), wherein a value corresponding to the amplitude of the predetermined input signal is corrected based on the asymmetry.
(3)
The correction unit,
Correcting the predetermined input signal based on the modulation factor when it is assumed that the predetermined input signal is converted into a PWM signal so that the waveform within one carrier frequency is symmetric. (1) or (2) An audio processing device according to claim 1.
(4)
The correction unit,
A value corresponding to the amplitude of the predetermined input signal is increased based on a modulation factor when it is assumed that the predetermined input signal is converted into a PWM signal so that a waveform within one carrier frequency is symmetric. The audio processing device according to any one of (1) to (3).
(5)
An analysis unit that analyzes a driving capability of an amplification unit that drives the PWM signal converted by the PWM conversion unit;
The correction unit,
The audio processing device according to any one of (1) to (4), wherein the predetermined input signal is corrected according to the driving capability analyzed by the analysis unit.
(6)
The analysis unit,
Analyzing the pulse width of the PWM signal allowed by the amplification unit as the driving capability,
The correction unit,
A modulation factor is calculated based on the pulse width analyzed by the analyzer, assuming that the predetermined input signal is converted into a PWM signal so that a waveform within one carrier frequency is symmetric, and the calculated modulation factor is calculated. The audio processing device according to (5), wherein the predetermined input signal is corrected based on the following.
(7)
The correction unit,
When the pulse width analyzed by the analyzer satisfies a predetermined condition, the predetermined input signal is not corrected,
The PWM converter includes:
The audio processing device according to (5) or (6), which converts a predetermined input signal before being corrected by the correction unit into a PWM signal.
(8)
The audio processing device according to any one of (1) to (7), wherein the predetermined input signal is a signal obtained by converting a signal in a direct stream digital (DSD) format into a pulse code modulation (PCM) format.
(9)
Computer
Correcting the predetermined input signal based on the asymmetry of the waveform within one carrier frequency when the predetermined input signal is converted into a PWM (Pulse Width Modulation) signal;
An audio processing method for converting a corrected predetermined input signal into a PWM signal.
(10)
Computer
A correction unit that corrects the predetermined input signal based on the asymmetry of the waveform within one carrier frequency when the predetermined input signal is converted into a PWM (Pulse Width Modulation) signal;
An audio processing program for functioning as a PWM conversion unit that converts a predetermined input signal corrected by the correction unit into a PWM signal.

1, 2 audio processing system 10 audio source 20

audio output device

100, 100A audio processing device 110 DSP
111 Low-pass filter 112 Correction unit 120 ΔΣ modulator 130 PWM conversion unit 140 Amplification unit 150 Analysis unit

Claims

A correction unit that corrects the predetermined input signal based on the asymmetry of the waveform within one carrier frequency when the predetermined input signal is converted into a PWM (Pulse Width Modulation) signal;
A PWM conversion unit that converts a predetermined input signal corrected by the correction unit into a PWM signal.
The correction unit,
The audio processing device according to claim 1, wherein a value corresponding to the amplitude of the predetermined input signal is corrected based on the asymmetry.
The correction unit,
The audio processing apparatus according to claim 1, wherein the predetermined input signal is corrected based on a modulation factor when it is assumed that the predetermined input signal is converted into a PWM signal so that a waveform within one carrier frequency is symmetric. .
The correction unit,
A value corresponding to the amplitude of the predetermined input signal is increased based on a modulation factor when it is assumed that the predetermined input signal is converted into a PWM signal so that a waveform within one carrier frequency is symmetric. An audio processing device according to claim 1.
An analysis unit that analyzes a driving capability of an amplification unit that drives the PWM signal converted by the PWM conversion unit;
The correction unit,
The audio processing device according to claim 1, wherein the predetermined input signal is corrected according to the driving ability analyzed by the analysis unit.
The analysis unit,
Analyzing the pulse width of the PWM signal allowed by the amplification unit as the driving capability,
The correction unit,
A modulation factor is calculated based on the pulse width analyzed by the analyzer, assuming that the predetermined input signal is converted into a PWM signal so that a waveform within one carrier frequency is symmetric, and the calculated modulation factor is calculated. The audio processing device according to claim 5, wherein the predetermined input signal is corrected based on the following.
The correction unit,
When the pulse width analyzed by the analyzer satisfies a predetermined condition, the predetermined input signal is not corrected,
The PWM converter includes:
The audio processing device according to claim 5, wherein a predetermined input signal before being corrected by the correction unit is converted into a PWM signal.
The audio processing device according to claim 1, wherein the predetermined input signal is a signal obtained by converting a signal in a DSD (Direct Stream Digital) format into a PCM (Pulse Code Modulation) format.
Computer
Correcting the predetermined input signal based on the asymmetry of the waveform within one carrier frequency when the predetermined input signal is converted into a PWM (Pulse Width Modulation) signal;
An audio processing method for converting a corrected predetermined input signal into a PWM signal.
Computer
A correction unit that corrects the predetermined input signal based on the asymmetry of the waveform within one carrier frequency when the predetermined input signal is converted into a PWM (Pulse Width Modulation) signal;
An audio processing program for functioning as a PWM conversion unit that converts a predetermined input signal corrected by the correction unit into a PWM signal.