WO2015125195A1 - オーディオ信号増幅装置 - Google Patents
オーディオ信号増幅装置 Download PDFInfo
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- WO2015125195A1 WO2015125195A1 PCT/JP2014/005914 JP2014005914W WO2015125195A1 WO 2015125195 A1 WO2015125195 A1 WO 2015125195A1 JP 2014005914 W JP2014005914 W JP 2014005914W WO 2015125195 A1 WO2015125195 A1 WO 2015125195A1
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- signal
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- correction
- audio signal
- pass filter
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R3/00—Circuits for transducers, loudspeakers or microphones
- H04R3/04—Circuits for transducers, loudspeakers or microphones for correcting frequency response
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Processing of the speech or voice signal to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0208—Noise filtering
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- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03F—AMPLIFIERS
- H03F1/00—Details of amplifiers with only discharge tubes, only semiconductor devices or only unspecified devices as amplifying elements
- H03F1/34—Negative-feedback-circuit arrangements with or without positive feedback
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- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03F—AMPLIFIERS
- H03F3/00—Amplifiers with only discharge tubes or only semiconductor devices as amplifying elements
- H03F3/181—Low frequency amplifiers, e.g. audio preamplifiers
- H03F3/183—Low frequency amplifiers, e.g. audio preamplifiers with semiconductor devices only
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- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03F—AMPLIFIERS
- H03F3/00—Amplifiers with only discharge tubes or only semiconductor devices as amplifying elements
- H03F3/20—Power amplifiers, e.g. Class B amplifiers, Class C amplifiers
- H03F3/21—Power amplifiers, e.g. Class B amplifiers, Class C amplifiers with semiconductor devices only
- H03F3/217—Class D power amplifiers; Switching amplifiers
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- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03F—AMPLIFIERS
- H03F3/00—Amplifiers with only discharge tubes or only semiconductor devices as amplifying elements
- H03F3/20—Power amplifiers, e.g. Class B amplifiers, Class C amplifiers
- H03F3/21—Power amplifiers, e.g. Class B amplifiers, Class C amplifiers with semiconductor devices only
- H03F3/217—Class D power amplifiers; Switching amplifiers
- H03F3/2175—Class D power amplifiers; Switching amplifiers using analogue-digital or digital-analogue conversion
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- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03F—AMPLIFIERS
- H03F2200/00—Indexing scheme relating to amplifiers
- H03F2200/03—Indexing scheme relating to amplifiers the amplifier being designed for audio applications
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- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03F—AMPLIFIERS
- H03F2200/00—Indexing scheme relating to amplifiers
- H03F2200/171—A filter circuit coupled to the output of an amplifier
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- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03F—AMPLIFIERS
- H03F2200/00—Indexing scheme relating to amplifiers
- H03F2200/331—Sigma delta modulation being used in an amplifying circuit
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- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03F—AMPLIFIERS
- H03F2200/00—Indexing scheme relating to amplifiers
- H03F2200/351—Pulse width modulation being used in an amplifying circuit
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- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03F—AMPLIFIERS
- H03F2200/00—Indexing scheme relating to amplifiers
- H03F2200/432—Two or more amplifiers of different type are coupled in parallel at the input or output, e.g. a class D and a linear amplifier, a class B and a class A amplifier
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- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03F—AMPLIFIERS
- H03F3/00—Amplifiers with only discharge tubes or only semiconductor devices as amplifying elements
- H03F3/20—Power amplifiers, e.g. Class B amplifiers, Class C amplifiers
- H03F3/21—Power amplifiers, e.g. Class B amplifiers, Class C amplifiers with semiconductor devices only
- H03F3/217—Class D power amplifiers; Switching amplifiers
- H03F3/2171—Class D power amplifiers; Switching amplifiers with field-effect devices
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- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03F—AMPLIFIERS
- H03F3/00—Amplifiers with only discharge tubes or only semiconductor devices as amplifying elements
- H03F3/20—Power amplifiers, e.g. Class B amplifiers, Class C amplifiers
- H03F3/21—Power amplifiers, e.g. Class B amplifiers, Class C amplifiers with semiconductor devices only
- H03F3/217—Class D power amplifiers; Switching amplifiers
- H03F3/2173—Class D power amplifiers; Switching amplifiers of the bridge type
Definitions
- the present disclosure relates to an audio signal amplifying apparatus.
- Patent Document 1 discloses a class D power amplifier as a signal amplifier.
- the present disclosure provides an audio signal amplifying apparatus that realizes high-quality audio signals.
- An audio signal amplifying apparatus includes a delta-sigma modulation unit that resamples an input digital audio signal with a quantization number smaller than the quantization number of the digital audio signal, and an amplitude direction of an output signal of the delta-sigma modulation unit
- a pulse width modulation unit that converts a gradation into a pulse width modulation signal having a gradation of the pulse width
- a power amplification unit that amplifies the output signal of the pulse width modulation unit, and a predetermined cutoff of the output signals of the power amplification unit
- a low-pass filter that reduces and outputs components higher than the frequency
- a correction processing unit that generates a correction signal for correcting the digital audio signal.
- the correction processing unit includes a switch that turns on and off the connection with the low-pass filter, and generates a correction signal by connecting a speaker to the low-pass filter when the switch is on.
- the audio signal amplifying apparatus can achieve high sound quality.
- FIG. 1 is a block diagram of the audio signal amplifying apparatus according to the first embodiment.
- FIG. 2 is a block diagram showing the delta-sigma modulation unit in the first embodiment.
- FIG. 3 is a diagram illustrating the noise floor characteristics of the input signal and the noise floor characteristics of the output signal of the noise shaper according to the first embodiment.
- FIG. 4 is a diagram showing corrected frequency amplitude characteristics and frequency phase characteristics in the first embodiment.
- FIG. 5A is a diagram illustrating characteristics of a conventional low-pass filter.
- FIG. 5B is a diagram showing the characteristics of the low-pass filter in the first embodiment.
- FIG. 6 is a diagram showing the noise floor characteristics, frequency amplitude characteristics, and frequency phase characteristics of the output signal of the delta-sigma modulation section in the first embodiment.
- FIG. 1 is a block diagram of the audio signal amplifying apparatus according to the first embodiment.
- FIG. 2 is a block diagram showing the delta-sigma modulation unit in the first embodiment.
- FIG. 3 is
- FIG. 7 is a block diagram of a conventional class D power amplifier.
- FIG. 8 is a diagram for explaining the problem of the frequency characteristics of the low-pass filter.
- FIG. 9A is a frequency characteristic diagram of a low-pass filter when a pure resistor is connected as a load.
- FIG. 9B is a frequency characteristic diagram of the low-pass filter when a speaker is connected as a load.
- FIG. 10 is a diagram illustrating the impedance with respect to the frequency of the output signal of the speaker.
- FIG. 11 is a block diagram of a conventional audio signal amplifying apparatus.
- FIG. 7 is a block diagram of a conventional class D power amplifier that receives a digital audio signal.
- the class D power amplifier 101 receives a digital signal from an input terminal 102.
- the class D power amplifier 101 includes a delta-sigma modulation unit 103, a pulse width modulation unit 104, a power amplification unit 105, a low-pass filter 106, and a speaker 107.
- the delta-sigma modulation unit 103 secures a dynamic range determined by the quantization number of the digital signal input from the input terminal 102 within a predetermined band, for example, up to 20 kHz that is an audible band, and reduces the quantization number.
- the pulse width modulation unit 104 converts the gradation in the amplitude direction of the output signal of the delta sigma modulation unit 103 into a pulse width.
- the power amplifying unit 105 amplifies the binary signal that is the output of the pulse width modulation unit 104.
- the low-pass filter 106 removes the carrier signal generated by the pulse width modulation unit 104 and the requantization noise generated by the delta-sigma modulation unit 103 from the output of the power amplification unit 105, and extracts the audio signal.
- the low-pass filter 106 includes a coil and a capacitor in order to reduce power loss.
- the class D power amplifier 101 converts the audio signal input to the input terminal 102 into a binary signal and amplifies the signal amplitude to near the power supply voltage, so that power loss in the amplifying element is not limited. Get smaller and get higher power efficiency.
- FIG. 8 is a diagram for explaining the problem of the frequency characteristics of the low-pass filter 106.
- the low pass filter 106 includes a coil 106a and a capacitor 106b.
- a load 801 is connected to the output of the low-pass filter 106.
- FIG. 9A is a frequency characteristic diagram of the low-pass filter 106 when a pure resistor is connected as the load 801
- FIG. 9B is a frequency characteristic diagram of the low-pass filter 106 when the speaker 107 is connected as the load 801.
- the horizontal axis indicates the frequency of the output signal of the low-pass filter 106
- the vertical axis indicates the output level of the frequency amplitude characteristic and the frequency phase characteristic.
- the low-pass filter 106 is designed so as to obtain frequency amplitude characteristics and frequency phase characteristics as shown in FIG. 9A.
- FIG. 10 is a diagram showing the impedance with respect to the frequency of the output signal of the speaker 107.
- the impedance of the speaker 107 that is, the resistance value changes according to the frequency. Therefore, when the speaker 107 is connected as the load 801, the frequency characteristic of the output of the low-pass filter 106 is as shown in FIG. 9B.
- the frequency characteristics are significantly different from the frequency characteristics shown in FIG. 9A, as shown in FIG. 9B.
- FIG. 11 is a block diagram of a conventional audio signal amplifying apparatus.
- the audio signal amplifying apparatus 1100 includes an input terminal 102, a class D power amplifier 101, an NFB (Negative FeedBack) 1101, and an adder 1102.
- the output signal of the low-pass filter 106 is negatively fed back to the adder 1102 via the NFB 1101.
- the adder 1102 amplified, and again a class D power amplifier. 101.
- the negative feedback by the NFB 1101 is always used, so that distortion generated in the low-pass filter 106 in the audio signal amplifying apparatus 1100 is suppressed.
- the phase margin indicating how much the phase difference at a frequency of 0 dB is 0 degree, that is, how much the phase amount is 180 degrees above. It is known that the amplitude margin indicating how much the amplitude at a frequency of 60 degrees or more and a phase difference of 0 degrees is lower than 0 dB is 6 dB or more.
- the speaker 107 to be connected changes in amplitude and phase with respect to the change in frequency, and the combination of changes varies depending on the type of the speaker 107 to be connected. Therefore, it is not easy to satisfy the conditions for a stable negative feedback configuration.
- an A / D converter is required to negatively feed back the generated error with a digital signal, and a time delay based on the operation of the A / D converter is generated. Negative feedback is not possible. Even if the negative feedback can be correctly performed, the distortion due to the phase generated in the low-pass filter 106 cannot be corrected.
- an audio signal amplifying apparatus using class D power amplification has a problem that the original characteristics of a speaker cannot be reproduced, that is, sound quality deteriorates due to the following phenomenon.
- FIG. 1 is a block diagram of an audio signal amplifying apparatus 100 according to the first embodiment.
- the audio signal amplifying apparatus 100 includes an input terminal 102, a class D power amplifier 101, and a correction processing unit 108.
- the correction processing unit 108 includes a switch 109, a response signal measurement unit 110, a correction unit 111, and a synthesis unit 112.
- the audio signal amplifying apparatus 100 receives a digital audio signal from the input terminal 102.
- the correction processing unit 108 performs a process of generating a correction signal described later only when the switch 109 is turned on.
- the switch 109 is turned on, for example, when the user uses the audio signal amplifying apparatus 100 for the first time. Then, when the processing of the correction processing unit 108 is completed, the switch 109 is turned off.
- the audio signal amplifying apparatus 100 is provided with a button (not shown) for the user to turn on the switch 109.
- This button is provided with a lamp, and the lamp is turned on while the switch 109 is pressed ON, and the lamp is turned off when the switch 109 is turned OFF, so that the user can visually recognize the state of the switch. .
- the response signal measuring unit 110 measures the frequency amplitude characteristic and the frequency phase characteristic of the class D power amplifier 101.
- the correction unit 111 generates a correction signal for correcting the frequency amplitude characteristic and the frequency phase characteristic of the class D power amplifier 101 based on the measurement result of the response signal measurement unit 110.
- the synthesizing unit 112 stores the generated correction signal, and corrects the frequency amplitude characteristic and the frequency phase characteristic of the digital audio signal that is the input signal of the input terminal 102 based on the stored correction signal.
- the class D power amplifier 101 converts an input digital audio signal, for example, a 16-bit digital audio signal into a 1-bit binary signal in the case of a compact disc, performs power amplification, and is not required by the low-pass filter 106.
- the audio signal with the amplified power is extracted by removing the unnecessary band.
- FIG. 2 is a block diagram of the delta-sigma modulation unit 103.
- the delta sigma modulation unit 103 includes an oversampling filter 201 and a noise shaper 202.
- the input signal is 16 bits and the output signal is 3 bits.
- the combination of the input signal and the output signal is not limited to this.
- the input signal may be M bits (M is an integer equal to or greater than 1) and the output signal may be N bits (N is an integer greater than 1 satisfying M> N).
- M is an integer equal to or greater than 1
- N is an integer greater than 1 satisfying M> N.
- specific values of M are 16, 20, and 24.
- the oversampling filter 201 converts the sampling frequency of the digital audio signal, which is an input signal, to a power of 2, and removes the aliasing component from the signal.
- the noise shaper 202 requantizes the oversampled digital audio signal with a smaller quantization number than the input signal.
- the noise shaper 202 reduces the requantization noise generated when requantization, that is, the noise floor, in an audio band, for example, 20 kHz or less.
- FIG. 3 is a diagram illustrating the noise floor characteristic of the input signal and the noise floor characteristic of the output signal of the noise shaper 202.
- the left side of FIG. 3 is the noise floor characteristic of the input signal of the noise shaper 202, and the right side is the noise floor characteristic of the output signal of the noise shaper 202.
- the horizontal axis is the signal frequency, and the vertical axis is the output level of the noise floor.
- the noise floor of the input signal is constant, but the noise floor of the output signal is relatively small below 20 kHz and relatively large above 20 kHz.
- the pulse width modulation unit 104 converts the gradation in the amplitude direction of the output signal of the delta sigma modulation unit 103 into a pulse width modulation signal having the gradation of the pulse width.
- the 16-bit input signal is converted into a binary pulse width modulation signal by the processing of the delta sigma modulation unit 103 and the pulse width modulation unit 104.
- the power amplification unit 105 amplifies power with respect to the amplitude of the pulse width modulation signal.
- the power amplifying unit 105 amplifies the input binary signal to almost the power supply voltage and reduces the reactive voltage, thereby reducing the power loss.
- the low-pass filter 106 includes a coil 106a and a capacitor 106b as described with reference to FIG. 8 in order to suppress power loss. Then, on the assumption that a pure resistance is connected to the output of the low-pass filter 106 as a load 801, a characteristic of the low-pass filter 106, for example, a Butterworth characteristic is determined. This characteristic is generally designed so that the characteristic is flat in the pass band, and the requantization noise generated in the carrier wave of the pulse width modulation unit 104 and the delta-sigma modulation unit 103 is attenuated in the stop band.
- correction processing unit 108 Next, the correction processing unit 108 will be described.
- the correction processing unit 108 generates a correction signal only when the switch 109 is turned on.
- the correction processing unit 108 measures the output signal of the low-pass filter 106 with the speaker 107 connected.
- a measurement signal is input to the input terminal 102 with the speaker 107 connected.
- a TSP (Time Stretched Pulse) signal having impulse characteristics is used as the measurement signal.
- the measurement signal is not limited to the TSP signal, and an impulse signal or another signal may be used as the measurement signal.
- the measurement range is a predetermined audio band, for example, 20 kHz or less.
- the response signal measurement unit 110 measures the output signal of the low-pass filter 106 with respect to the input measurement signal.
- the response signal measurement unit 110 measures the impulse response of the class D power amplifier 101.
- the response signal measurement unit 110 performs fast Fourier transform (FFT) processing on the measured output signal to obtain the frequency amplitude characteristic and the frequency phase characteristic of the class D power amplifier 101.
- FFT fast Fourier transform
- the TSP signal is a signal obtained by scanning (sweeping) a sine wave whose phase is proportional to the square of the frequency at a high speed from a low frequency low range to a high frequency range in the frequency range to be measured.
- the response signal measurement unit 110 calculates an impulse response from the measured output signal by analyzing an inverse function.
- the correcting unit 111 calculates a ratio between the frequency amplitude characteristic obtained by the response signal measuring unit 110 and the frequency amplitude characteristic targeted by the audio signal amplifying apparatus 100.
- the correction unit 111 calculates a correction signal related to the frequency amplitude characteristic from this ratio.
- the calculation of the correction signal by the correction unit 111 is preferably determined by the balance between the correction effect and the correction processing amount. That is, a higher correction effect is expected with a simple correction process.
- the correction unit 111 determines the correction content in consideration of the subjective characteristics of the person viewing the audio signal. For example, a frequency characteristic that a human feels in his / her ear is equivalent. In this case, if the octave analysis is used, the correction unit 111 can increase the subjective effect and reduce the processing amount.
- the correction unit 111 divides the frequency amplitude characteristic obtained by the response signal measurement unit 110 into 1 / N octave bands (N is a positive integer), and calculates an average value for each divided band. Then, using the calculated average value, a value at a frequency point, which is actually processed in the correction process, is calculated and used as a frequency amplitude measurement value for calculating a correction signal. Similarly, a target frequency amplitude characteristic value is also calculated. Then, the correction signal is calculated by taking the ratio of the two.
- the correction unit 111 when calculating the correction signal of the frequency amplitude characteristic, the correction unit 111 weights a specific octave band in consideration of the human auditory characteristic, the result obtained by original subjective evaluation, and the like.
- the response signal measurement unit 110 obtains the frequency phase characteristic by performing FFT on the measured impulse response. However, since the trigonometric function used in FFT has periodicity, the frequency phase characteristic may take a discontinuous value. The response signal measurement unit 110 calculates the frequency phase characteristic considering this. The response signal measurement unit 110 calculates a phase delay characteristic obtained by dividing the frequency phase characteristic by the angular frequency, or a group delay characteristic obtained by differentiating the frequency phase characteristic by the angular frequency.
- the difference between the calculated phase delay characteristic or group delay characteristic and the phase delay characteristic or group delay characteristic targeted by the audio signal amplifying apparatus 100 is calculated, and the inverse process of the procedure for calculating the phase delay characteristic or group delay phase characteristic is performed.
- the frequency phase characteristic for reducing the difference from the target value is calculated.
- the frequency phase characteristic at the frequency point that is actually processed by the correction process is extracted, and a correction signal of the frequency phase characteristic is calculated.
- the combining unit 112 stores the correction signal generated by the correction unit 111.
- the synthesizer 112 synthesizes the correction signal with the input digital audio signal.
- a convolution operation is performed by FIR (Finite Impulse Response). That is, the calculated correction signal is subjected to inverse FFT to calculate an impulse response and convolved with the input signal.
- FIR Finite Impulse Response
- the synthesis method is not limited to this.
- the input digital audio signal may be subjected to FFT, multiplied by the correction signal, and the result may be subjected to inverse FFT.
- the audio signal amplifying apparatus 100 switches the switch 109 of the correction processing unit 108 to OFF.
- FIG. 4 is a diagram showing the corrected frequency amplitude characteristic and frequency phase characteristic.
- the audio signal amplification apparatus 100 approaches the target frequency amplitude characteristic and frequency phase characteristic.
- the audio signal amplification device 100 includes the delta sigma modulation unit 103 that resamples the input digital audio signal with a quantization number smaller than the quantization number of the digital audio signal, and the delta sigma modulation unit.
- a pulse width modulation unit 104 that converts a gradation in the amplitude direction of the output signal 103 into a pulse width modulation signal having a gradation of the pulse width; a power amplification unit 105 that amplifies the output signal of the pulse width modulation unit 104; A low-pass filter 106 that reduces and outputs a component higher than a predetermined cut-off frequency in the output signal of the power amplification unit 105 and a correction processing unit 108 that generates a correction signal for correcting the digital audio signal are provided.
- the correction processing unit 108 includes a switch 109 that turns on and off the connection with the low-pass filter 106. When the switch 109 is on, a correction signal is generated by connecting a speaker to the low-pass filter 106.
- the correction processing unit 108 of the audio signal amplifying device 100 of the present disclosure includes a response signal measuring unit 110 that measures the output signal of the low-pass filter 106, and a correction that generates a correction signal based on the measurement result of the response signal measuring unit 110. And a combining unit 112 that combines the correction signal with the digital audio signal.
- the audio signal amplifying apparatus 100 when the switch 109 is on, a measurement signal having an impulse characteristic is input to the delta-sigma modulation unit 103, and the correction unit 111 determines from the measurement result of the response signal measurement unit 110. A correction signal for the frequency amplitude characteristic or a correction signal for the frequency phase characteristic is generated.
- the class D power amplifier 101 of the audio signal amplifying apparatus 100 efficiently amplifies the power and supplies power to the load. Audio signals closer to not only the characteristics but also the frequency phase characteristics can be output.
- the response signal measurement unit 110 measures both the frequency amplitude characteristic and the frequency phase characteristic of the signal output from the low-pass filter 106, but the present invention is not limited to this. One of the wave number amplitude characteristic and the frequency phase characteristic may be measured.
- the audio signal amplifying apparatus 100 may be configured as one module, or the class D power amplifier 101 and the correction processing unit 108 may be configured as separate modules.
- the class D power amplifier 101 is used as the power amplifier, but the present invention is not limited to this.
- a class A power amplifier, a class AB power amplifier, a class B power amplifier, or the like may be used.
- the delta-sigma modulation unit 103 performs processing to increase the influence of re-quantization noise accompanying the reduction in the number of quantization in a frequency region higher than the audio band as described with reference to FIG.
- the speaker 107 is connected to the low-pass filter 106, characteristics as a target filter cannot be obtained, noise cannot be sufficiently attenuated, and sound quality may be deteriorated.
- FIG. 5A is a diagram showing the characteristics of the conventional low-pass filter 106
- FIG. 5B is a diagram showing the characteristics of the low-pass filter 106 of the present embodiment.
- the horizontal axis represents the frequency of the input signal of the low-pass filter 106
- the vertical axis represents the output level of the low-pass filter 106. Comparing FIG. 5A and FIG. 5B, FIG. 5B shows that the cutoff frequency of the low-pass filter 106 is lower than FIG. 5A.
- the cutoff frequency of the low-pass filter 106 is a frequency when the gain of the output signal decreases by a predetermined value or more with respect to an input signal having a predetermined frequency or higher.
- the low-pass filter 106 When an input signal having a frequency equal to or higher than the cutoff frequency is input, the low-pass filter 106 suppresses and outputs a signal having a higher cutoff frequency. That is, the low-pass filter 106 reduces and outputs an input signal having a cutoff frequency or higher.
- FIG. 6 is a diagram showing the noise floor characteristic, frequency amplitude characteristic, and frequency phase characteristic of the output signal of the delta-sigma modulation unit 103.
- the left side of FIG. 6 is a characteristic when the characteristic of the conventional low-pass filter 106 is used, and the left side of FIG. 6 is a characteristic when the characteristic of the low-pass filter 106 of the present embodiment is used.
- the horizontal axis represents the frequency of the output signal of the low-pass filter 106, and the vertical axis represents the output level. In FIG.
- the frequency amplitude characteristic, frequency phase characteristic and noise of the output signal of the low-pass filter 106 are compared.
- the response decreases as the frequency of the output signal of the low-pass filter 106 increases. Thereby, requantization noise can be reduced.
- the audio signal amplifying device of the present disclosure can be applied to a sound reproducing device that requires high sound quality.
Abstract
Description
[1-1.課題]
まず、オーディオ信号増幅装置の高音質化における課題を説明する。図7は、従来のデジタルオーディオ信号を入力とするD級電力増幅器のブロック図である。D級電力増幅器101は、入力端子102からデジタル信号が入力される。D級電力増幅器101は、デルタシグマ変調部103、パルス幅変調部104、電力増幅部105、ローパスフィルタ106とスピーカ107で構成される。
図1は、実施の形態1におけるオーディオ信号増幅装置100のブロック図である。オーディオ信号増幅装置100は、入力端子102と、D級電力増幅器101と、補正処理部108で構成される。補正処理部108は、スイッチ109と、応答信号測定部110と、補正部111と、合成部112で構成される。
まず、D級電力増幅器101について説明する。D級電力増幅器101は、入力されるデジタルオーディオ信号、例えばコンパクトディスクの場合であれば16ビットのデジタルオーディオ信号を1ビットの2値信号に変換して、電力増幅を行い、ローパスフィルタ106で不要な帯域を除去して電力増幅されたオーディオ信号を取り出す。
次に、補正処理部108について説明する。補正処理部108は、スイッチ109をONにした場合にのみ補正信号の生成する処理を行う。
応答信号測定部110は、入力した計測用信号に対するローパスフィルタ106の出力信号を測定する。応答信号測定部110は、D級電力増幅器101のインパルス応答を測定する。応答信号測定部110は、測定した出力信号に高速フーリエ変換(FFT:Fast Fourier Transform)処理を施し、D級電力増幅器101の周波数振幅特性と周波数位相特性を求める。TSP信号は、位相が周波数の二乗に比例した正弦波を、計測対象とする周波数範囲における低周波低域から高周波帯域に周波数を高速に走査(スイープ)した信号である。TSP信号は、単一のパルス信号と比べてエネルギーが大きいため、出力信号を複数回同期加算すると、大きなS/N(Signal-Noise ratio)を得やすい。応答信号測定部110は、測定した出力信号から逆関数の解析によりインパルス応答を算出する。
補正部111は、応答信号測定部110で求めた周波数振幅特性と、オーディオ信号増幅装置100が目標とする周波数振幅特性との比を算出する。補正部111は、この比から周波数振幅特性に関する補正信号を算出する。
応答信号測定部110は、測定したインパルス応答をFFTすることで周波数位相特性を求める。しかし、FFTで用いられる三角関数は周期性を有するので、周波数位相特性は、不連続な値をとることがある。応答信号測定部110は、これを考慮して周波数位相特性を算出する。応答信号測定部110は、周波数位相特性を角周波数で除算した位相遅延特性、または周波数位相特性を角周波数で微分した群遅延特性を算出する。そして、算出した位相遅延特性または群遅延特性と、オーディオ信号増幅装置100が目標とする位相遅延特性または群遅延特性の差分を算出し、位相遅延特性または群遅延位相特性を算出した手順の逆処理を行い、目標値との差分を小さくするための周波数位相特性を算出する。そして、補正処理で実際に処理する、周波数点での周波数位相特性を抽出して、周波数位相特性の補正信号を算出する。
合成部112は、補正部111が生成した補正信号を記憶する。合成部112は、入力されるデジタルオーディオ信号に補正信号を合成する。合成には、FIR(Finite Impulse Response)による畳込み演算を行う。即ち、算出した補正信号を逆FFTして、インパルス応答を算出し、入力信号に畳込む処理を行う。
図4は、補正後の周波数振幅特性および周波数位相特性を示す図である。図9Bの周波数振幅特性および周波数位相特性を補正すると図4に示すように、オーディオ信号増幅装置100が目標とする周波数振幅特性、周波数位相特性に近づく。
次に、ローパスフィルタ106の遮断周波数について説明する。
101 D級電力増幅器
102 入力端子
103 デルタシグマ変調部
104 パルス幅変調部
105 電力増幅部
106 ローパスフィルタ
106a コイル
106b コンデンサ
107 スピーカ
108 補正処理部
109 スイッチ
110 応答信号測定部
111 補正部
112 合成部
1101 NFB
1102 加算器
Claims (3)
- 入力されるデジタルオーディオ信号を前記デジタルオーディオ信号の量子化数よりも小さい量子化数で再サンプリングするデルタシグマ変調部と、
前記デルタシグマ変調部の出力信号の振幅方向の階調をパルスの幅の諧調とするパルス幅変調信号に変換するパルス幅変調部と、
前記パルス幅変調部の出力信号を電力増幅する電力増幅部と、
前記電力増幅部の出力信号のうち所定の遮断周波数より高い成分を逓減させて出力するローパスフィルタと、
前記デジタルオーディオ信号を補正する補正信号を生成する補正処理部と、を備え、
前記補正処理部は、前記ローパスフィルタとの接続をオンオフするスイッチを含み、前記スイッチがオンの時に前記ローパスフィルタにスピーカを接続して前記補正信号を生成する、
オーディオ信号増幅装置。 - 前記補正処理部は、
前記ローパスフィルタの出力信号を測定する応答信号測定部と、
前記応答信号測定部の測定結果に基づいて前記補正信号を生成する補正部と、
前記補正信号を前記デジタルオーディオ信号に合成する合成部と、を含む、
請求項1に記載のオーディオ信号増幅装置。 - 前記スイッチがオンの場合、前記デルタシグマ変調部にインパルス特性を有する計測用信号が入力され、
前記補正部は、前記応答信号測定部の測定結果から周波数振幅特性に対する補正信号または周波数位相特性に対する補正信号を生成する、
請求項1に記載のオーディオ信号増幅装置。
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Citations (3)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
JP2002159088A (ja) * | 2000-11-17 | 2002-05-31 | Kenwood Corp | 増幅器、及び音声出力方法 |
JP2003133959A (ja) * | 2001-10-29 | 2003-05-09 | Sony Corp | D/a変換器および出力増幅回路 |
JP2006121529A (ja) * | 2004-10-22 | 2006-05-11 | Pioneer Electronic Corp | D級増幅装置、増幅制御プログラム及び情報記録媒体 |
Family Cites Families (8)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US6414614B1 (en) * | 1999-02-23 | 2002-07-02 | Cirrus Logic, Inc. | Power output stage compensation for digital output amplifiers |
KR20060027637A (ko) * | 2004-09-23 | 2006-03-28 | 주식회사 팬택앤큐리텔 | 무선통신 단말기에서의 오디오 선형 증폭 장치 및 그 방법 |
JP4802765B2 (ja) * | 2005-03-18 | 2011-10-26 | ヤマハ株式会社 | D級増幅器 |
US8116368B2 (en) | 2006-07-27 | 2012-02-14 | National University Corporation Nagoya Institute Of Technology | PWM signal generator, PWM signal generating device, and digital amplifier |
JP5321263B2 (ja) * | 2009-06-12 | 2013-10-23 | ソニー株式会社 | 信号処理装置、信号処理方法 |
KR20110036371A (ko) * | 2009-10-01 | 2011-04-07 | 삼성전자주식회사 | 오디오 증폭기 |
JP6044269B2 (ja) * | 2011-11-04 | 2016-12-14 | ヤマハ株式会社 | 自励発振型d級アンプおよび自励発振型d級アンプの自励発振周波数制御方法 |
JP6151619B2 (ja) * | 2013-10-07 | 2017-06-21 | クラリオン株式会社 | 音場測定装置、音場測定方法および音場測定プログラム |
-
2014
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Patent Citations (3)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
JP2002159088A (ja) * | 2000-11-17 | 2002-05-31 | Kenwood Corp | 増幅器、及び音声出力方法 |
JP2003133959A (ja) * | 2001-10-29 | 2003-05-09 | Sony Corp | D/a変換器および出力増幅回路 |
JP2006121529A (ja) * | 2004-10-22 | 2006-05-11 | Pioneer Electronic Corp | D級増幅装置、増幅制御プログラム及び情報記録媒体 |
Non-Patent Citations (1)
Title |
---|
See also references of EP3110004A4 * |
Cited By (2)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US11955935B2 (en) | 2019-08-27 | 2024-04-09 | Panasonic Intellectual Property Management Co., Ltd. | Signal processing device and adjusting method |
JP7344606B1 (ja) | 2022-12-19 | 2023-09-14 | 純一 角元 | 1ビット符号化信号の積和演算手段とその応用の畳込演算手段と積分変換手段と信号抽出手段とアナログ情報検索システム |
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