WO2020040068A1 - Dispositif de traitement du son, procédé de traitement du son et programme de traitement du son - Google Patents

Dispositif de traitement du son, procédé de traitement du son et programme de traitement du son Download PDF

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Publication number
WO2020040068A1
WO2020040068A1 PCT/JP2019/032200 JP2019032200W WO2020040068A1 WO 2020040068 A1 WO2020040068 A1 WO 2020040068A1 JP 2019032200 W JP2019032200 W JP 2019032200W WO 2020040068 A1 WO2020040068 A1 WO 2020040068A1
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signal
pwm
waveform
input signal
predetermined input
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PCT/JP2019/032200
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English (en)
Japanese (ja)
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宜紀 田森
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ソニー株式会社
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03M—CODING; DECODING; CODE CONVERSION IN GENERAL
    • H03M3/00—Conversion of analogue values to or from differential modulation
    • H03M3/02—Delta modulation, i.e. one-bit differential modulation

Definitions

  • 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.
  • a high-resolution sound source which are audio data with a sound quality exceeding that of music CDs.
  • a signal recorded in a DSD (Direct Stream Digital) format is used.
  • 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”).
  • PWM Pulse Width Modulation
  • a so-called digital amplifier that generates a drive signal by switching is often used.
  • 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.
  • 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.
  • the pulse width of the input PWM signal may be limited due to the nature of the amplifier (amplifier circuit).
  • a signal having a narrow pulse width less than the limit hereinafter, referred to as “narrow pulse”
  • noise may be generated or a device may be adversely affected.
  • a high-resolution sound source has a very high sampling frequency, and thus tends to easily generate a narrow pulse.
  • a signal having a somewhat wide pulse width in which generation of a narrow pulse is suppressed is generated.
  • 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.
  • 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.
  • an audio processing device 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
  • 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. 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 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.
  • 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.
  • 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.
  • DSD signal an audio signal recorded in the DSD format
  • the DSD signal is a ⁇ modulated PDM (PulseulDensity Modulation) signal.
  • 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.
  • Fs 44.1 kHz
  • an audio signal such as a DSD signal may be represented by a combination of a sampling frequency and a bit depth.
  • 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.
  • 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. .
  • the audio processing apparatus 100 includes a DSP (Digital Signal Processor) 110, a ⁇ modulator 120, a PWM conversion unit 130, and an amplification unit 140.
  • DSP Digital Signal Processor
  • 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).
  • 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.
  • 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 ⁇ modulator 120 (step S13).
  • the ⁇ 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 ⁇ 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 ⁇ 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.
  • 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.
  • 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.
  • a signal corresponding to “1” in the DSD signal is represented by PMW (+)
  • a signal corresponding to “0” in the DSD signal is represented by PMW ( ⁇ ). Called.
  • the waveform 38 indicated by “1” corresponds to PMW (+)
  • the waveform 36 indicated by "0” corresponds to PMW (-).
  • 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.
  • the master clock is “1024Fs”.
  • width 32 corresponds to a sampling frequency (carrier frequency)
  • width 34 corresponds to a slot.
  • 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.
  • PWM (+) occupies 16 slots and PWM (-) occupies 0 slot.
  • PWM (+) occupies 16 slots and PWM (-) occupies 0 slot.
  • 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.
  • 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.
  • PWM (+) maintains a line-symmetric shape when PWM (-) has a width of 2 slots
  • PWM (+) will have a width of 14 slots.
  • 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.
  • 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.
  • 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.
  • 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.
  • PWM (+) maintains a line-symmetric shape when PWM ( ⁇ ) has a width of 4 slots
  • PWM (+) has a width of 12 slots by nature.
  • 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.
  • 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).
  • PWM (+) also has 4 slots if it is desired to maintain a line-symmetric shape.
  • 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.
  • the sound processing device 100 generates the waveform 50 illustrated in FIG. 2C.
  • the waveform 56 indicating PWM (-) has a pulse width of 4 slots to avoid a narrow pulse.
  • 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.
  • 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.
  • 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.
  • 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.
  • 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.
  • 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.
  • the waveform 62 shows the original DSD signal held by the audio source 10 as a sine wave.
  • 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.
  • the waveform 61 has different absolute values of the upper and lower amplitudes because the PWM ( ⁇ ) and the PWM (+) are asymmetric.
  • the level on the plus side is “0.5”
  • the level on the minus side is “ ⁇ (0.5 ⁇ N (N is a predetermined level determined by the degree of asymmetry. Number)) ".
  • 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.
  • 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.
  • 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.
  • the waveform 63 in FIG. 3 is a sine wave corresponding to the input signal after the above-described correction processing.
  • 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.
  • the audio processing device 100 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.
  • 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.
  • 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.
  • 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").
  • 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.
  • the pulse width of PWM (+) is “14/16” and the pulse width of PWM ( ⁇ ) is “2/16”.
  • 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.
  • the waveform 50 has a PWM (+) pulse width of “14/16” and a PWM ( ⁇ ) pulse width of “4/16”.
  • the amplitude of the corresponding sine wave 67 has different absolute values on the plus side and the minus side.
  • 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.
  • the pulse width of the PWM (+) of the waveform 50 is “14/16”
  • the pulse width of the temporary PWM ( ⁇ ) is "2/16”.
  • 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.
  • 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”.
  • the sine wave 67 is an asymmetric waveform having a positive amplitude “0.375” and a negative amplitude “ ⁇ 0.25”.
  • FIG. 5 is a diagram (2) illustrating a correction process according to the present disclosure.
  • 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.
  • 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.
  • 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.
  • 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.
  • the sine wave 67 has “0.375” on the plus side and “ ⁇ 0.0. 25 ".
  • 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.
  • 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.
  • 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 correction unit 112 corrects the bit string of the input signal of [64 FS, 16 bits] through the low-pass filter 111.
  • 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 ⁇ 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.
  • the sine wave 72 corresponding to the waveform 51 has the same absolute value on the plus side and the minus side.
  • 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.
  • 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.
  • 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.
  • 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.
  • 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.
  • 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.
  • the audio processing device 100 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.
  • the audio processing according to the present disclosure is realized by a configuration in which the DSP 110 and the ⁇ modulator 120 are installed before the PWM conversion unit 130, and thus does not depend on the performance of the amplification unit 140.
  • 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.
  • 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.
  • 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.
  • 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.
  • 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.
  • Step S101 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).
  • 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 ⁇ 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).
  • the amplifier 140 generates an output signal from the PWM signal (step S106).
  • 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.
  • 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.
  • 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.
  • 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.
  • FIG. 7 is a diagram illustrating a configuration example of the audio processing system 2 according to the second embodiment of the present disclosure.
  • 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.
  • description of the same configuration as that according to the first embodiment will be omitted.
  • 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.
  • the degree of modulation is calculated and the DSP 110 performs correction based on the calculated degree of modulation.
  • the audio processing device 100A obtains variables such as the characteristics of the above-described amplifying unit 140 and analyzes those values to enable dynamic correction processing.
  • 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
  • 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.
  • 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.
  • 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).
  • 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.
  • 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.
  • 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.
  • 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).
  • 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.
  • 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.
  • 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.
  • 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).
  • 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.
  • 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.
  • FIG. 8 is a flowchart illustrating a flow of a process according to the second embodiment of the present disclosure.
  • 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).
  • 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.
  • Step S202 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).
  • 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 ⁇ 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.
  • the amplifier 140 generates an output signal from the PWM signal (step S207).
  • 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.
  • FIG. 9 is a diagram illustrating an example of a waveform of a PWM signal according to a modified example of the present disclosure.
  • a waveform 81 shown in FIG. 9 shows an example of a PWM signal waveform in the balance driving method.
  • the PWM signal has symmetry at the center within one carrier frequency.
  • a waveform 81 a waveform obtained by differentially combining a positive signal and a negative signal has symmetry.
  • 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.
  • 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.
  • 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.
  • the waveform of the differential combination of PWM (+ P) and PWM (+ N) in one carrier are asymmetric.
  • 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.
  • 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.
  • 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.
  • 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.
  • BIOS Basic Input Output System
  • 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.
  • 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).
  • 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.
  • the CPU 1100 receives data from an input device such as a keyboard and a mouse via the input / output interface 1600.
  • the CPU 1100 transmits data to an output device such as a display, a speaker, or a printer via the input / output interface 1600.
  • 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.
  • 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.
  • 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.
  • 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.
  • the CPU 1100 reads and executes the program data 1450 from the HDD 1400.
  • the CPU 1100 may acquire these programs from another device via the external network 1550.
  • 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 (1), wherein a value corresponding to the amplitude of the predetermined input signal is corrected based on the asymmetry.
  • 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 An audio processing device according to claim 1.
  • 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).
  • 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.
  • 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.
  • 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.
  • 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.
  • PCM pulse code modulation
  • 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 ⁇ modulator 130 PWM conversion unit 140 Amplification unit 150 Analysis unit

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  • Engineering & Computer Science (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Theoretical Computer Science (AREA)
  • Amplifiers (AREA)

Abstract

Un dispositif de traitement du son (100) comprend : une unité de correction (112) qui, sur la base d'une asymétrie d'une forme d'onde dans une fréquence porteuse lorsqu'un signal d'entrée prédéterminé est converti en un signal de modulation de largeur d'impulsion (MLI), corrige le signal d'entrée prédéterminé ; et une unité de conversion MLI (130) qui convertit, en un signal MLI, le signal d'entrée prédéterminé ayant été corrigé par l'unité de correction (112).
PCT/JP2019/032200 2018-08-24 2019-08-16 Dispositif de traitement du son, procédé de traitement du son et programme de traitement du son WO2020040068A1 (fr)

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Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2003030373A1 (fr) * 2001-09-28 2003-04-10 Sony Corporation Appareil de modulation delta-sigma et appareil d'amplification de signaux
JP2003110376A (ja) * 2001-09-28 2003-04-11 Sony Corp 信号増幅装置
US20100329482A1 (en) * 2009-06-26 2010-12-30 Lee Yong-Hee Audio digital to analog converter and audio processing apparatus including the same

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2003030373A1 (fr) * 2001-09-28 2003-04-10 Sony Corporation Appareil de modulation delta-sigma et appareil d'amplification de signaux
JP2003110376A (ja) * 2001-09-28 2003-04-11 Sony Corp 信号増幅装置
US20100329482A1 (en) * 2009-06-26 2010-12-30 Lee Yong-Hee Audio digital to analog converter and audio processing apparatus including the same

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