WO2013183185A1 - 周波数特性変形装置 - Google Patents
周波数特性変形装置 Download PDFInfo
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- WO2013183185A1 WO2013183185A1 PCT/JP2012/082546 JP2012082546W WO2013183185A1 WO 2013183185 A1 WO2013183185 A1 WO 2013183185A1 JP 2012082546 W JP2012082546 W JP 2012082546W WO 2013183185 A1 WO2013183185 A1 WO 2013183185A1
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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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- 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
- G10L21/0216—Noise filtering characterised by the method used for estimating noise
- G10L21/0232—Processing in the frequency domain
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- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03G—CONTROL OF AMPLIFICATION
- H03G5/00—Tone control or bandwidth control in amplifiers
- H03G5/16—Automatic control
- H03G5/165—Equalizers; Volume or gain control in limited frequency bands
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- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03G—CONTROL OF AMPLIFICATION
- H03G9/00—Combinations of two or more types of control, e.g. gain control and tone control
- H03G9/005—Combinations of two or more types of control, e.g. gain control and tone control of digital or coded signals
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- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03G—CONTROL OF AMPLIFICATION
- H03G9/00—Combinations of two or more types of control, e.g. gain control and tone control
- H03G9/02—Combinations of two or more types of control, e.g. gain control and tone control in untuned amplifiers
- H03G9/025—Combinations of two or more types of control, e.g. gain control and tone control in untuned amplifiers frequency-dependent volume compression or expansion, e.g. multiple-band systems
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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
Definitions
- This invention relates to a signal processing technique for improving distortion and sound cracking in acoustic signal reproduction.
- the sound quality may deteriorate due to distortion or sound cracking.
- the first case can be described as follows.
- In recent acoustic signal reproduction systems there are an increasing number of devices for correcting frequency characteristics and adjusting volume by digital processing.
- the vertical axis represents the amplitude intensity of the digital signal
- the horizontal axis represents the frequency.
- a region where the signal is saturated and sound cracking occurs is shown in gray, and the boundary is shown by a bold line.
- Reference numerals 201, 202, and 203 denote examples of frequency characteristics of digital audio signals whose frequency characteristics are corrected, where 201 is a characteristic when the volume value is small, 202 is a characteristic when the volume value is medium, and 203 is This is a characteristic when the volume value is large.
- the volume values of 201 and 202 the sound signal does not exceed 0 dBFS, so that sound is not broken and the sound can be enjoyed with the original sound quality.
- the volume is increased as in 203, the signal strength of a part of the high frequency component exceeds 0 dBFS and is digitally saturated. When the signal is saturated, distortion and sound cracking occur and sound quality deteriorates.
- FIG. 3 shows the displacement width of the speaker diaphragm when the signal (V) is changed and only the signal frequency is changed and input to the speaker.
- the displacement width of the speaker diaphragm is substantially constant at a frequency component lower than F0 (the lowest resonance frequency of the speaker), and the displacement width is about ⁇ 12 dB / oct at a frequency component higher than F0. Decrease. This indicates that the speaker diaphragm swings with a larger displacement width when a low frequency component below F0 is input to the speaker than when a high frequency component is input. Therefore, if a signal containing a large amount of low frequency components is input to the speaker and the voltage is increased, the maximum displacement width of the diaphragm will be exceeded at a certain voltage or higher.
- the vertical axis indicates the amplitude intensity of the signal
- the horizontal axis indicates the frequency.
- the region where the sound cracking occurs beyond the displacement limit of the speaker diaphragm is shown in gray, and the boundary is shown in bold.
- the displacement limit of the speaker diaphragm is an inclination of +12 dB / oct.
- 401, 402, and 403 indicate frequency characteristics of an acoustic signal reproduced by a speaker, and a case including a lot of low frequency components is assumed.
- 401 is a characteristic when the volume value is small
- 402 is a characteristic when the volume value is medium
- 403 is a frequency characteristic when the volume value is large. Since the sound signal containing a large amount of low frequency components does not exceed the maximum displacement width of the speaker diaphragm, the sound can be enjoyed with the original sound quality without being broken, even if the sound signal contains a lot of low frequency components. . However, if the volume is increased as in 402 and 403, the maximum displacement width of the speaker diaphragm is exceeded, so that distortion and sound cracking occur and sound quality deteriorates.
- Distortion and sound cracking are sounds that are not included in the original acoustic signal, and thus become a major impediment when trying to enjoy music.
- the signal 1303 is output after passing through an HPF (High Pass Filter) 1302 that suppresses low frequency components with respect to the input signal 1301.
- HPF High Pass Filter
- the conventional technique has a problem that the original sound quality cannot be enjoyed by excessively suppressing the low frequency component in order to prevent sound cracking when driving at a high voltage (at a high volume).
- FIG. 18 is a processing block of the amplitude limiting device disclosed in Patent Document 1.
- the amplitude limit for suppressing an excessive input the amount of distortion due to the amplitude limit characteristic is detected, and the gain for each frequency band is controlled based on this value, so that the sound quality deterioration due to the amplitude limit is reduced. It is relaxed.
- the technique suppresses the intensity of the entire signal component of 0 to 100 Hz, components other than the frequency component to be suppressed (components of 60 to 100 Hz) are also suppressed.
- the displacement width of the speaker amplitude plate has frequency characteristics.
- the amplitude limiting device disclosed in Patent Document 1 has a processing configuration that reflects the frequency characteristics of the displacement width. Absent. For this reason, it can be said that it does not have a function to prevent sound cracking that occurs due to exceeding the maximum displacement of the speaker diaphragm.
- the present invention has been made to solve the above-described problems, and an object of the present invention is to provide a frequency characteristic deforming device capable of preventing distortion and sound cracking in speaker reproduction while maintaining sound quality.
- a frequency characteristic modification device includes a filter that deforms a frequency characteristic of a target signal, a phase correction unit that corrects the phase characteristic of the target signal and substantially equals the phase characteristic of the filter, and a phase correction
- a first multiplier for adjusting the gain of the signal output from the unit
- a second multiplier for adjusting the gain of the signal output from the filter
- a gain coefficient of the first multiplier and a second multiplier
- a coefficient determination unit that determines the gain coefficients of the first and second multipliers so that the sum of the gain coefficients of the two becomes a constant value, and two signals output from the first multiplier and the second multiplier And an adder for adding.
- FIG. 2 is a diagram illustrating the principle of a frequency characteristic modification device according to Embodiment 1.
- FIG. It is a figure which shows the relationship between the amplitude limit of a digital signal, and the frequency characteristic of a sound source. It is a figure which shows the displacement characteristic of a speaker diaphragm. It is a figure which shows the relationship between the vibration limit of a speaker, and the frequency characteristic of a sound source.
- FIG. 6 is an explanatory diagram showing transition of frequency characteristics due to two gains of the frequency characteristic modifying device according to the first embodiment.
- It is a principle explanatory view of a frequency characteristic modification device according to a second embodiment.
- FIG. 10 is an explanatory diagram showing transition of frequency characteristics due to two gains of the frequency characteristic modification device according to the second embodiment.
- FIG. 10 is an explanatory diagram showing transition of frequency characteristics due to three gains of the frequency characteristic modifying device according to the third embodiment. It is a principle explanatory view of a frequency characteristic modification device according to Embodiment 4. It is a figure which shows the Example of the low-frequency extraction part of the frequency characteristic transformation apparatus by Embodiment 4. It is a figure which shows the other Example of the low-frequency extraction part of the frequency characteristic transformation apparatus by Embodiment 4. It is a figure which shows the addition image of the harmonic according to the low-pass attenuation
- FIG. FIG. 10 is a diagram illustrating the principle of a frequency characteristic modification device according to a fifth embodiment.
- FIG. 10 is a principle explanatory diagram of a frequency characteristic modifying device according to a sixth embodiment. It is principle explanatory drawing of a prior art. It is a processing block diagram of a prior art amplitude limiting device.
- FIG. FIG. 1 is a diagram showing an embodiment of the present invention.
- the input signal 101 input to the frequency characteristic modification apparatus according to the present invention is branched and sent to the phase correction unit 701 and the HPF 702.
- the phase correction unit 701 corrects only the phase characteristic so as to be substantially the same as the phase characteristic of the HPF 702 without changing the frequency amplitude characteristic of the input signal, and the obtained signal 703 is excessively larger than the first multiplier 705.
- the HPF 702 filters the input signal 101 and outputs the obtained signal 704 to the second multiplier 706 and the excessive input estimation unit 102.
- phase correction unit 701 that corrects the phase so as to have substantially the same phase characteristics as the HPF 702
- HPF 702 is realized by one stage of the second-order IIR filter
- the phase characteristic rotates just 90 degrees at the cutoff frequency, and gradually rotates up to 180 degrees in the frequency components thereafter.
- the phase correction unit 701 that realizes such phase characteristics can be configured by an all-pass filter using a first-order IIR filter.
- the phase characteristic rotates just 180 degrees at the cutoff frequency, and gradually rotates up to 360 degrees in the frequency components thereafter.
- the phase correction unit that realizes such phase characteristics can be configured by an all-pass filter using a secondary IIR filter.
- phase correction unit 701 can be configured by sample delay processing. In this way, the phase correction unit 701 having the same phase characteristics as the HPF 702 can be realized.
- the excessive input estimation unit 102 includes a speaker diaphragm displacement estimation unit 501.
- the speaker diaphragm displacement estimation unit 501 estimates the speaker diaphragm displacement value when the signal 703 is reproduced using information 502 such as the volume value and F0 of the target speaker, and the first speaker diaphragm displacement value 707. Ask for.
- the displacement value of the speaker diaphragm when the signal 704 is reproduced is estimated, and the second speaker diaphragm displacement value 708 is obtained.
- an LPF using a second-order IIR filter with F0 as a cut-off frequency is prepared, and this is multiplied by a volume value after passing through an input signal, so that the value is roughly proportional to the displacement width of the target speaker. Is required.
- the Q value can be changed in the LPF using the second-order IIR filter, the estimation accuracy can be improved by changing the Q value according to the degree of braking of the target speaker.
- the diaphragm displacement characteristics of the target speaker may be simulated by other methods such as an FIR filter.
- the two speaker diaphragm displacement values 707 and 708 thus obtained are output to the control unit 105.
- FIG. 5 shows two gains when the HPF 702 is realized with two stages of a second-order IIR filter with a cutoff frequency of 80 Hz, and the phase correction unit 701 is realized with one stage of a second-order IIR all-pass filter with a cutoff frequency of 80 Hz. It shows the transition of frequency characteristics by coefficient.
- the speaker diaphragm displacement value 707 is X1
- the speaker diaphragm displacement value 708 is X2
- the gain coefficient for X1 is A1
- the gain coefficient for X2 is A2
- T it can be realized by obtaining A1 and A2 that satisfy the following expression (1).
- A1 + A2 1
- ABS (x) represents the absolute value of x.
- A1 is a signal based on a signal obtained by correcting only the phase characteristics, and the closer the A1 is to 1, the less the frequency characteristics are deformed.
- A2 may be adopted.
- A1 thus obtained is output to the first multiplier 705 as a gain coefficient 709. Also, A2 is output to the second multiplier 706 as a gain coefficient 710.
- the first multiplier 705 multiplies the input signal 703 and the gain coefficient 709 and outputs the obtained signal 711 toward the adder 713.
- the second multiplier 706 multiplies the input signal 704 and the gain coefficient 710 and outputs the obtained signal 712 toward the adder 713.
- the adder 713 adds the two input signals 711 and 712 and outputs the obtained signal as an output signal 107.
- the processing configuration of the first embodiment can prevent the reproduced sound signal from being excessively input. For this reason, the effect that distortion and sound cracking can be suppressed is obtained by the present invention. Further, by making the cut-off frequency as low as possible by the control unit, it is possible to obtain an effect that distortion and sound cracking can be prevented with a minimum necessary frequency characteristic change.
- Embodiment 2 By replacing the HPF 702 described in the first embodiment with an LPF, the frequency characteristic of the signal is adjusted so that the maximum amplitude of the digital signal is not exceeded in the digital acoustic signal that has been subjected to the frequency characteristic correction in which the correction of the high frequency component is often performed. It is also possible to deform.
- FIG. 6 shows a processing configuration showing an embodiment when the HPF 702 is replaced with an LPF.
- the input signal 101 input to the frequency characteristic modification apparatus according to the present invention is branched and sent to the phase correction unit 701 and the LPF 901.
- the phase correction unit 701 corrects only the phase characteristic so as to be substantially the same as the phase characteristic of the LPF 901 without changing the frequency amplitude characteristic of the input signal, and the obtained signal 703 is excessively larger than the first multiplier 705. Output to the input estimation unit 102.
- the LPF 901 filters the input signal 101 and outputs the obtained signal 902 to the second multiplier 706 and the excessive input estimation unit 102.
- the phase correction unit 701 can be realized by an all-pass filter or a sample delay process as in the first embodiment, detailed description thereof is omitted.
- the excessive input estimation unit 102 includes a digital signal amplitude calculation unit 601.
- the digital signal amplitude calculation unit 601 multiplies the volume value 602 and the input signal 703 to obtain the first amplitude value 707.
- the second amplitude value 708 is obtained by multiplying the volume value 602 and the input signal 902.
- the two amplitude values 707 and 708 thus obtained are output to the control unit 105.
- the predetermined threshold value is normally set to 0 dBFS, but is not limited to this. If the speaker input resistance is not matched with the amplifier output and the amplifier output is to be limited, a value smaller than this is set. May be set.
- FIG. 7 shows two gains when the LPF 702 is realized by two stages of a second-order IIR filter having a cutoff frequency of 6000 Hz and the phase correction unit 701 is realized by one stage of a second-order IIR all-pass filter having a cutoff frequency of 6000 Hz. It shows the transition of frequency characteristics by coefficient.
- components having the same phase are added at a ratio of 1 in total, so that flat characteristics can be maintained without increasing or decreasing the intensity. Since a specific method for calculating the two gain coefficients is obtained in the same manner as in the first embodiment, the description thereof is omitted.
- A1 thus obtained is output to the first multiplier 705 as a gain coefficient 709. Also, A2 is output to the second multiplier 706 as a gain coefficient 710.
- the first multiplier 705 multiplies the input signal 703 and the gain coefficient 709 and outputs the obtained signal 711 toward the adder 713.
- the second multiplier 706 multiplies the input signal 902 and the gain coefficient 710 and outputs the obtained signal 712 to the adder 713.
- the adder 713 adds the two input signals 711 and 712 and outputs the obtained signal as an output signal 107.
- the processing configuration of the second embodiment it is possible to suppress the amplitude value of the digital acoustic signal in which many high-frequency components are corrected, and to suppress distortion and sound cracking. Is obtained.
- the control unit of this embodiment can make the cut-off frequency of the LPF as large as possible, an effect that distortion and sound cracking can be prevented by a change in the minimum frequency characteristics is also obtained.
- the frequency characteristic deforming unit is realized by one phase correction unit and one HPF or LPF.
- the present invention is not limited to this, and the frequency is formed by a plurality of phase correction units and a plurality of HPFs or LPFs. You may implement
- FIG. 8 is a diagram illustrating an example in which a frequency characteristic deforming unit is realized by three phase correction units and three HPFs. Hereinafter, the operation of this embodiment will be described.
- the input signal 101 input to the frequency characteristic modification apparatus according to the present invention is branched into three and sent to the first HPF 1101, the second HPF 1102, and the third HPF 1103.
- the first HPF 1101 filters the input signal and outputs the obtained signal 1104 toward the first phase correction unit 1107.
- the second HPF 1102 filters the input signal and outputs the obtained signal 1105 toward the second phase correction unit 1108.
- the third HPF 1103 filters the input signal and outputs the obtained signal 1106 toward the third phase correction unit 1109.
- the phase characteristic is set so as to be substantially the same as the phase characteristic when both the second HPF 1102 and the third HPF 1103 are processed.
- the corrected signal 1110 is output to the first multiplier 1113 and the excessive input estimation unit 102.
- the second phase correction unit 1108 without changing the frequency amplitude characteristic of the signal, only the phase characteristic is set so as to be substantially the same as the phase characteristic when both the first HPF 1101 and the third HPF 1103 are processed.
- the corrected signal 1111 is output to the second multiplier 1114 and the excessive input estimation unit 102.
- the third phase correction unit 1109 does not change the frequency amplitude characteristic of the signal, but only the phase characteristic so as to be substantially the same as the phase characteristic when both the first HPF 1101 and the second HPF 1102 are processed.
- the corrected signal 1112 is output to the third multiplier 1115 and the excessive input estimation unit 102.
- the excessive input estimation unit 102 includes a speaker diaphragm displacement estimation unit 501.
- the speaker diaphragm displacement estimation unit 501 estimates the displacement value of the speaker diaphragm when the signal 1110 is reproduced using information 502 such as the volume value and F0 of the target speaker, and the first speaker diaphragm displacement value 1116. Ask for.
- the displacement value of the speaker diaphragm when the signal 1111 is reproduced is estimated, and the second speaker diaphragm displacement value 1117 is obtained.
- the displacement value of the speaker diaphragm when the signal 1112 is reproduced is estimated, and the third speaker diaphragm displacement value 1118 is obtained. Since a specific example of the displacement value estimation is obtained by the same method as in the first embodiment, detailed description thereof is omitted.
- the three speaker diaphragm displacement values 1116, 1117, and 1118 thus obtained are output to the control unit 105.
- the control unit 105 when the input three speaker diaphragm displacement values 1116, 1117, and 1118 are multiplied by different gain coefficients and added, the absolute value of the amplitude value falls within a predetermined threshold value. Three gain factors are obtained. However, the sum of the three gain coefficients is 1.
- FIG. 9 shows that the first HPF 1101 is realized by two stages of a secondary IIR filter having a cutoff frequency of 30 Hz, the second HPF 1102 is realized by two stages of a secondary IIR filter having a cutoff frequency of 70 Hz, and a third HPF 1103 is realized. Is realized with four stages of a secondary IIR filter having a cutoff frequency of 140 Hz, and the first phase correction unit 1107 includes one stage of a secondary IIR filter having a cutoff frequency of 70 Hz and two stages of a secondary IIR filter having a cutoff frequency of 140 Hz.
- the second phase correction unit 1108 is realized by connecting a second-order IIR filter with a cutoff frequency of 30 Hz and a second-order IIR filter with a cutoff frequency of 140 Hz in series.
- the third phase correction unit 1109 includes a second-order IIR filter having a cutoff frequency of 30 Hz and a second-order IIR filter having a cutoff frequency of 70 Hz. It represents the transition of the frequency characteristics due to the three gain factors when implemented by connecting the 1-stage IR filter in series.
- the speaker diaphragm displacement value 1116 is X1
- the speaker diaphragm displacement value 1117 is X2
- the speaker diaphragm displacement value 1118 is X3, and the gain coefficient for X1 is set.
- A2 is the gain coefficient for A1 and X2
- A3 is the gain coefficient for X3
- T is the predetermined threshold
- A1 obtained in this way is output to the first multiplier 1113 as a gain coefficient 1119. Further, A2 is output to the second multiplier 1114 as the gain coefficient 1120. Further, A3 is output to the third multiplier 1115 as a gain coefficient 1121.
- the first multiplier 1113 multiplies the input signal 1110 and the gain coefficient 1119 and outputs the obtained signal 1122 toward the adder 713.
- the second multiplier 1114 multiplies the input signal 1111 and the gain coefficient 1120 and outputs the obtained signal 1123 toward the adder 713.
- the third multiplier 1115 multiplies the input signal 1112 and the gain coefficient 1121 and outputs the obtained signal 1124 to the adder 713.
- the adder 713 adds the three input signals 1122, 1123, and 1124 and outputs the obtained signal as the output signal 107.
- the frequency characteristic deforming unit can be realized by three phase corrections and three HPFs.
- similar to is realizable is acquired.
- by increasing the number of phase corrections and HPFs it becomes possible to realize characteristics closer to those of normal HPFs.
- the frequency characteristics of the signal are modified so that the maximum amplitude of the digital signal is not exceeded in the digital acoustic signal that has been subjected to frequency characteristic correction that often corrects high frequency components. It is also possible to make it.
- FIG. 10 is a diagram showing another embodiment of the present invention.
- the input signal 101 input to the signal processing apparatus according to the present invention is branched and sent to the phase correction unit 701 and the HPF 702.
- the phase correction unit 701 corrects only the phase characteristic so as to be substantially the same as the phase characteristic of the HPF 702 without changing the frequency amplitude characteristic of the input signal, and the obtained signal 703 is overloaded with the first multiplier 705. It outputs toward the input estimation part 102 and the harmonic signal generation part 2001.
- the HPF 702 filters the input signal 101 and outputs the obtained signal 704 to the adder 2003.
- the phase correction unit can be realized by an all-pass filter or a sample delay process as in the first embodiment, detailed description thereof is omitted.
- the harmonic signal generation unit 2001 includes a low-frequency extraction unit 2004, a harmonic generation unit 2006, and a multiplier 2008.
- the low-frequency extraction unit 2004 receives the output signal 703 from the phase correction unit 701, extracts a low frequency that is cut by the HPF 702, and outputs the obtained signal 2005 to the harmonic generation unit 2006.
- a method of realizing the low-frequency extraction unit 2004 that extracts a low-frequency band cut by the HPF 702 is configured from a subtractor 2011 as shown in FIG.
- the harmonic generation unit 2006 generates harmonics of the output signal 2005 of the low-frequency extraction unit 2004 up to the nth order (n is an integer of 3 or more), and outputs the obtained signal 2007 to the multiplier 2008.
- the harmonic generation unit 2006 can be realized by waveform modification such as peak hold, full-wave rectification, half-wave rectification, multiplication by m times (m is an integer) of the signal 2005, frequency division, and the like. Any device that generates both harmonics and even harmonics may be used.
- the multiplier 2008 multiplies the output signal 2007 of the harmonic generation unit 2006 by a user-preferred gain coefficient, and outputs the obtained signal 2002 to the adder 2003.
- the gain coefficient multiplied by the multiplier 2008 is changed according to the user's preference by preparing a plurality of fixed gain coefficients in advance.
- the adder 2003 adds the two input signals 704 and 2002, and outputs the obtained signal 2003 ′ toward the excessive input estimation unit 102 and the second multiplier 706.
- the excessive input estimation unit 102 includes a speaker diaphragm displacement estimation unit 501.
- the speaker diaphragm displacement estimation unit 501 estimates the speaker diaphragm displacement value when the signal 703 is reproduced using information 502 such as the volume value and F0 of the target speaker, and the first speaker diaphragm displacement value 707. Ask for.
- the displacement value of the speaker diaphragm when the signal 2003 ′ is reproduced is estimated, and the second speaker diaphragm displacement value 708 is obtained. Since a specific example of the displacement value estimation is obtained by the same method as in the first embodiment, detailed description thereof is omitted.
- the two speaker diaphragm displacement values 707 and 708 obtained by the speaker diaphragm displacement estimation unit 501 are output to the control unit 105.
- the control unit 105 when the input two speaker diaphragm displacement values 707 and 708 are multiplied by different gain coefficients and then added, the absolute value of the amplitude value falls within a predetermined threshold value. Find two gain factors. However, the sum of the two gain coefficients is 1. If the two gain coefficients are changed under such conditions, different low-frequency attenuation effects can be realized. Since a specific example of the low-frequency attenuation effect has the same characteristics as those of the first embodiment, detailed description thereof is omitted.
- FIG. 13 shows low-frequency attenuation when the HPF 702 is realized with two stages of a second-order IIR filter with a cutoff frequency of 80 Hz, and the phase correction unit 701 is realized with one stage of a second-order IIR all-pass filter with a cutoff frequency of 80 Hz. It shows the addition image of harmonics according to the effect.
- the HPF 702 is realized with two stages of a second-order IIR filter with a cutoff frequency of 80 Hz
- the phase correction unit 701 is realized with one stage of a second-order IIR all-pass filter with a cutoff frequency of 80 Hz. It shows the addition image of harmonics according to the effect.
- A1 obtained by the control unit 105 is output to the first multiplier 705 as a gain coefficient 709.
- A2 is output to the second multiplier 706 as a gain coefficient 710.
- the first multiplier 705 multiplies the input signal 703 and the gain coefficient 709 and outputs the obtained signal 711 toward the adder 713.
- the second multiplier 706 multiplies the input signal 2003 ′ by the gain coefficient 710 and outputs the obtained signal 712 toward the adder 713.
- the adder 713 adds the two input signals 711 and 712 and outputs the obtained signal as an output signal 107.
- low-frequency harmonics cut by the frequency characteristic deforming unit are generated up to the nth order in the harmonic signal generating unit (n is an integer of 3 or more), and psychoacoustic.
- the effect is that the low range cut by the feature “Missing fundamental” can be simulated.
- the control unit controls the output signal of the HPF and the gain of the generated harmonic signal together, the low-frequency interpolation effect can be changed according to the low-frequency attenuation characteristic.
- Missing fundamental is a feature that gives an illusion that when a sound of two or more frequencies is heard, a sound of the difference frequency is heard.
- FIG. 14 is a diagram illustrating an example in which a frequency characteristic deforming unit is realized by three phase correcting units and three HPFs.
- the input signal 101 input to the signal processing apparatus according to the present invention is branched into six, and is sent to the first HPF 1101, the second HPF 1102, the third HPF 1103, the first LPF 2101, the second LPF 2102, and the third LPF 2103. Sent.
- the first HPF 1101 filters the input signal and outputs the obtained signal 1104 toward the first phase correction unit 1107.
- the second HPF 1102 filters the input signal and outputs the obtained signal 1105 toward the second phase correction unit 1108.
- the third HPF 1103 filters the input signal and outputs the obtained signal 1106 toward the third phase correction unit 1109.
- the phase characteristic is set so as to be substantially the same as the phase characteristic when both the second HPF 1102 and the third HPF 1103 are processed.
- the corrected signal 1110 is output to the adder 2125.
- the second phase correction unit 1108 without changing the frequency amplitude characteristic of the signal, only the phase characteristic is set so as to be substantially the same as the phase characteristic when both the first HPF 1101 and the third HPF 1103 are processed.
- the corrected signal 1111 is output to the adder 2126.
- the third phase correction unit 1109 does not change the frequency amplitude characteristic of the signal, but only the phase characteristic so as to be substantially the same as the phase characteristic when both the first HPF 1101 and the second HPF 1102 are processed.
- the corrected signal 1112 is output to the adder 2127.
- each phase correction unit can be realized by an all-pass filter or a sample delay process as in the first embodiment, detailed description thereof is omitted.
- the first LPF 2101 processes the input signal 101 with a filter having the same filter specification as that of the first HPF 1101, and outputs the obtained signal 2104 toward the fourth phase correction unit 2107.
- the second LPF 2102 processes the input signal 101 with a filter having the same filter specification as that of the second HPF 1102, and outputs the obtained signal 2105 toward the fifth phase correction unit 2108.
- the third LPF 2103 processes the input signal 101 with a filter having the same filter specifications as the third HPF 1103, and outputs the obtained signal 2106 to the sixth phase correction unit 2109.
- phase characteristic is set so as to be substantially the same as the phase characteristic when both the second LPF 2102 and the third LPF 2103 are processed.
- the corrected signal 2110 is output to the first harmonic generation unit 2113.
- the fifth phase correction unit 2108 does not change the frequency amplitude characteristic of the signal, but only the phase characteristic so as to be substantially the same as the phase characteristic when both the first LPF 2101 and the third LPF 2103 are processed.
- the corrected signal 2111 is output to the second harmonic generation unit 2114.
- phase characteristic is set so as to be substantially the same as the phase characteristic when both the first LPF 2101 and the second LPF 2102 are processed without changing the frequency amplitude characteristic of the signal.
- the corrected signal 2112 is output to the third harmonic generation unit 2115.
- the first harmonic generation unit 2113 generates a harmonic of the signal 2110 and outputs the obtained signal 2116 toward the multiplier 2119.
- the second harmonic generation unit 2114 generates a harmonic of the signal 2111 and outputs the obtained signal 2117 to the multiplier 2120.
- the third harmonic generation unit 2115 generates a harmonic of the signal 2112 and outputs the obtained signal 2118 toward the multiplier 2121.
- each harmonic generation unit is generated by waveform deformation such as peak hold, full wave rectification, half wave rectification, m multiplication (m is an integer), frequency division, etc., and odd harmonics. Any device that generates both waves and even harmonics may be used.
- Multiplier 2119 multiplies signal 2116 by a user's favorite gain coefficient, and outputs the resulting signal 2122 to adder 2125.
- Multiplier 2120 multiplies signal 2117 by a user's favorite gain coefficient, and outputs the obtained signal 2123 to adder 2126.
- the multiplier 2121 multiplies the signal 2118 by a user's favorite gain coefficient and outputs the obtained signal 2124 to the adder 2127.
- the gain coefficient multiplied by the multipliers 2119, 2120, and 2121 is changed according to the user's preference by preparing a plurality of fixed gain coefficients in advance.
- the adder 2125 adds the two input signals 1110 and 2122 and outputs the obtained signal 2128 toward the excessive input estimation unit 102 and the first multiplier 1113.
- the adder 2126 adds the two input signals 1111 and 2123 and outputs the obtained signal 2129 toward the excessive input estimation unit 102 and the second multiplier 1114.
- the adder 2127 adds the two input signals 1112 and 2124 and outputs the obtained signal 2130 toward the excessive input estimation unit 102 and the third multiplier 1115.
- the excessive input estimation unit 102 includes a speaker diaphragm displacement estimation unit 501.
- the speaker diaphragm displacement estimation unit 501 estimates the speaker diaphragm displacement value when the signal 2128 is reproduced using information 502 such as the volume value and F0 of the target speaker, and the first speaker diaphragm displacement value 1116. Ask for.
- the displacement value of the speaker diaphragm when the signal 2129 is reproduced is estimated, and the second speaker diaphragm displacement value 1117 is obtained.
- the displacement value of the speaker diaphragm when the signal 2130 is reproduced is estimated, and the third speaker diaphragm displacement value 1118 is obtained. Since a specific example of the displacement value estimation is obtained by the same method as in the first embodiment, detailed description thereof is omitted.
- Three speaker diaphragm displacement values 1116, 1117, and 1118 obtained by the speaker diaphragm displacement estimation unit 501 are output to the control unit 105.
- the control unit 105 when the input three speaker diaphragm displacement values 1116, 1117, and 1118 are multiplied by different gain coefficients and added, the absolute value of the amplitude value falls within a predetermined threshold value. Three gain factors are obtained. However, the sum of the three gain coefficients is 1. If the three gain coefficients are changed under such conditions, different low-frequency attenuation effects can be realized. Since a specific example of the low-frequency attenuation effect has the same characteristics as in the third embodiment, detailed description thereof is omitted.
- the added amount of the cut low-frequency harmonics can be changed.
- the first HPF 1101 is realized with two stages of a second-order IIR filter with a cutoff frequency of 30 Hz
- the second HPF 1102 is realized with two stages of a second-order IIR filter with a cutoff frequency of 70 Hz
- the third HPF 1103 is realized.
- the first phase correction unit 1107 includes one stage of a secondary IIR filter having a cutoff frequency of 70 Hz and two stages of a secondary IIR filter having a cutoff frequency of 140 Hz.
- the second phase correction unit 1108 is realized by connecting a second-order IIR filter with a cutoff frequency of 30 Hz and a second-order IIR filter with a cutoff frequency of 140 Hz in series.
- the third phase correction unit 1109 includes a second-order IIR filter having a cutoff frequency of 30 Hz and a cutoff frequency of 70 Hz. It illustrates a sum image of the harmonic corresponding to the low-frequency damping effect when implemented by connecting the IIR filter 1 stage in series.
- A1 obtained by the control unit 105 is output to the first multiplier 1113 as the gain coefficient 1119. Further, A2 is output to the second multiplier 1114 as the gain coefficient 1120. Further, A3 is output to the third multiplier 1115 as a gain coefficient 1121.
- the first multiplier 1113 multiplies the input signal 2128 and the gain coefficient 1119 and outputs the obtained signal 1122 toward the adder 713.
- the second multiplier 1114 multiplies the input signal 2129 and the gain coefficient 1120 and outputs the obtained signal 1123 toward the adder 713.
- the third multiplier 1115 multiplies the input signal 2130 and the gain coefficient 1121 and outputs the obtained signal 1124 to the adder 713.
- the adder 713 adds the three input signals 1122, 1123, and 1124 and outputs the obtained signal as the output signal 107.
- the harmonic signal generation unit generates low-order harmonics cut by the frequency characteristic modification unit up to the nth order (n is an integer of 3 or more), and is psychoacoustic.
- n is an integer of 3 or more
- the effect that the low frequency cut by the feature “Missing fundamental” can be simulated is obtained.
- the control unit controls the output signal of the HPF and the gain of the generated harmonic signal together, the low-frequency interpolation effect can be changed according to the low-frequency attenuation characteristic.
- Embodiment 6 the output signal 2002 of the harmonic signal generation unit 2001 is added to the output signal of the HPF. However, the output signal 2002 of the harmonic signal generation unit 2001 is added to the subsequent stage of the frequency characteristic modification unit 103. You may make it the structure which adds.
- FIG. 16 is a diagram illustrating an example of a configuration in which the addition position of the output signal 2002 of the harmonic signal generation unit 2001 is changed to the subsequent stage of the frequency characteristic modification unit 103 in the fourth embodiment. Hereinafter, the operation of this embodiment will be described.
- the input signal 101 input to the signal processing apparatus according to the present invention is branched and sent to the phase correction unit 701 and the HPF 702.
- the phase correction unit 701 corrects only the phase characteristic so as to be substantially the same as the phase characteristic of the HPF 702 without changing the frequency amplitude characteristic of the input signal, and the obtained signal 703 is overloaded with the first multiplier 705. It outputs toward the input estimation part 102 and the harmonic signal generation part 2001.
- FIG. The HPF 702 filters the input signal 101 and outputs the obtained signal 704 to the multiplier 706 and the excessive input estimation unit 102.
- the phase correction unit can be realized by an all-pass filter or a sample delay process as in the first embodiment, detailed description thereof is omitted.
- the excessive input estimation unit 102 includes a speaker diaphragm displacement estimation unit 501.
- the speaker diaphragm displacement estimation unit 501 estimates the speaker diaphragm displacement value when the signal 703 is reproduced using information 502 such as the volume value and F0 of the target speaker, and the first speaker diaphragm displacement value 707. Ask for.
- the displacement value of the speaker diaphragm when the signal 704 is reproduced is estimated, and the second speaker diaphragm displacement value 708 is obtained. Since a specific example of the displacement value estimation is obtained by the same method as in the first embodiment, detailed description thereof is omitted.
- Two speaker diaphragm displacement values 707 and 708 obtained by the speaker diaphragm displacement estimation unit 501 are output to the control unit 105.
- the control unit 105 when the input two speaker diaphragm displacement values 707 and 708 are multiplied by different gain coefficients and then added, the absolute value of the amplitude value falls within a predetermined threshold value. Find two gain factors. However, the sum of the two gain coefficients is 1. If the two gain coefficients are changed under such conditions, different low-frequency attenuation effects can be realized. Since a specific example of the low-frequency attenuation effect has the same characteristics as those of the first embodiment, detailed description thereof is omitted. In FIG.
- A1 obtained by the control unit 105 is output to the first multiplier 705 as a gain coefficient 709. Also, A2 is output as gain coefficient 710 toward the second multiplier 706 and multiplier 2201.
- the first multiplier 705 multiplies the input signal 703 and the gain coefficient 709 and outputs the obtained signal 711 toward the adder 713.
- the second multiplier 706 multiplies the input signal 704 and the gain coefficient 710 and outputs the obtained signal 712 toward the adder 713.
- the adder 713 adds the two input signals 711 and 712, and outputs the obtained signal 2203 toward the addition 2204.
- the harmonic signal generation unit 2001 includes a low-frequency extraction unit 2004, a harmonic generation unit 2006, and a multiplier 2008.
- the low frequency extraction unit 2004 receives the output signal of the phase correction unit 701, extracts the low frequency that is cut by the HPF 702, and outputs the obtained signal 2005 to the harmonic generation unit 2006.
- description thereof is omitted.
- the harmonic generation unit 2006 generates harmonics of the output signal 2005 of the low-frequency extraction unit 2004 up to the nth order (n is an integer of 3 or more), and outputs the obtained signal 2007 to the multiplier 2008.
- the harmonic generation unit 2006 can be realized by waveform modification such as peak hold, full-wave rectification, half-wave rectification, multiplication by m times (m is an integer) of the signal 2005, frequency division, and the like. Any device that generates both harmonics and even harmonics may be used.
- the multiplier 2008 multiplies the output signal 2007 of the harmonic generation unit 2006 by a user-preferred gain coefficient, and outputs the obtained signal 2002 to the multiplier 2201.
- the gain coefficient multiplied by the multiplier 2008 is changed according to the user's preference by preparing a plurality of fixed gain coefficients in advance.
- Multiplier 2201 multiplies input signal 2002 and gain coefficient 710 and outputs the resulting signal 2202 to adder 2204.
- the multiplier 2201 can change the added amount of the cut low-frequency harmonics according to the low-frequency attenuation effect. Since the addition image of the harmonics according to the low-frequency attenuation effect is the same as that of the fourth embodiment, the description thereof is omitted.
- the adder 2204 adds the two input signals 2202 and 2203 and outputs the obtained signal as the output signal 107.
- the harmonic signal generation unit generates low-order harmonics cut by the frequency characteristic modification unit up to the nth order (n is an integer of 3 or more), and is psychoacoustic.
- n is an integer of 3 or more
- the effect that the low frequency cut by the feature “Missing fundamental” can be simulated is obtained.
- the control unit controls the output signal of the HPF and the gain of the generated harmonic signal together, the low-frequency interpolation effect can be changed according to the low-frequency attenuation characteristic.
- the frequency characteristic deforming apparatus can improve distortion and sound cracking in acoustic signal reproduction, and can be used for an audio reproducing apparatus.
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Abstract
Description
第一のケースについては、以下のように説明することができる。最近の音響信号再生システムではデジタル処理によって周波数特性を補正したり、音量を調整したりする装置が増えてきている。周波数特性の補正では、例えば高周波数成分を10dB増強すると、-10dBFS以上の音量値でデジタル信号が飽和する可能性がうまれる。なお0dBFSはデジタル信号の最大振幅値を表す。このため、大音量時に再生音がデジタル的に歪んでしまい、音質が劣化してしまうこととなる。この様子を図2に示す。
スピーカ再生では、スピーカの振動板が振れることのできる最大の変位幅があり、これを超えるような信号を入力するとスピーカ振動板がうまく振動することができなくなって歪みや音割れが発生する。ここで、スピーカ振動板の変位幅は入力信号の周波数に依存する。この関係を図3に示す。図3は、電圧(V)を変化させずに、信号の周波数のみを変化させてスピーカに入力したときにおけるスピーカ振動板の変位幅を示している。なお、図3は、実際にはスピーカの制動度合いを示すQ値等の違いによってF0近傍の特性が図3よりも膨らんだり平らになったりするが、おおまかな傾向は変わらない。また、変位幅の特性が図3に示す特性と異なっているスピーカに対しても本発明を適用することができるため、説明上の利点から、スピーカ振動板の変位幅の特性を図3と見做して以下の説明を行う。
図4において、縦軸は信号の振幅強度を、横軸は周波数を示す。また、スピーカ振動板の変位限界を超えて音割れの発生する領域をグレーで示し、その境界を太線で示す。ここで、図4の特性は音響信号の振幅値に対する特性なので、図3に示したスピーカの変位幅の特性とは異なり、スピーカ振動板の変位限界は+12dB/octの傾斜となる。
歪みや音割れは、本来の音響信号には含まれない音であるため、音楽を楽しもうとするときの大きな阻害要因となる。
実施の形態1.
図1は本発明の実施例を示す図である。以下、本実施例の動作について説明する。
本発明による周波数特性変形装置に入力された入力信号101は分岐され、位相補正部701とHPF702に送られる。
位相補正部701は入力信号の周波数振幅特性を変えずに、HPF702の位相特性と略同一の特性となるように位相特性のみを補正し、得られた信号703を第一の乗算器705と過大入力推定部102に向けて出力する。
HPF702は入力信号101をフィルタ処理し、得られた信号704を第二の乗算器706と過大入力推定部102に向けて出力する。
このように求めた2つのスピーカ振動板変位値707、708を制御部105へ向けて出力する。
このような条件で2つのゲイン係数を変えていくと、異なった低域減衰効果を実現することができる。図5は、HPF702を、カットオフ周波数80Hzの2次IIRフィルタ2段で実現し、位相補正部701を、カットオフ周波数80Hzの2次IIRの全域通過フィルタ1段で実現した時の2つのゲイン係数による周波数特性の遷移を表している。また、図5において、スピーカ振動板変位値707に対するゲイン係数をA1、スピーカ振動板変位値708に対するゲイン係数をA2としたとき、301はA1=1.0、A2=0.0の特性、302はA1=0.1、A2=0.9の特性、303は、A1=0.0、A2=1.0の特性を示す。このように、完全に平坦な特性(A1=1.0、A2=0.0)から、カットオフ周波数80Hzの2次IIRフィルタ2段と同じ特性(A1=0.0、A2=1.0)となる間で異なった低域減衰特性を実現できることがわかる。また、カットオフ周波数以上の周波数成分については、合計1となる割合で位相の揃った成分を加算するため、強度の増減はなく平坦な特性を保つことができる。
T>ABS(X1×A1+X2×A2) ・・・(1)
A1+A2=1
なお、ABS(x)はxの絶対値を表す。
第一の乗算器705では、入力された信号703とゲイン係数709を乗じて、得られた信号711を加算器713に向けて出力する。
第二の乗算器706では、入力された信号704とゲイン係数710を乗じて、得られた信号712を加算器713に向けて出力する。
加算器713では、入力された2つの信号711、712を加算し、得られた信号を出力信号107として出力する。
実施例1で説明したHPF702をLPFに置き換えることで、高周波数成分の補正が多いような周波数特性補正を施されたデジタル音響信号において、デジタル信号の最大振幅を超えないように信号の周波数特性を変形させることも可能である。図6はHPF702をLPFに置き換えた場合の実施例を示す処理構成である。
本発明による周波数特性変形装置に入力された入力信号101は分岐され、位相補正部701とLPF901に送られる。
位相補正部701は入力信号の周波数振幅特性を変えずに、LPF901の位相特性と略同一の特性となるように位相特性のみを補正し、得られた信号703を第一の乗算器705と過大入力推定部102に向けて出力する。
LPF901は入力信号101をフィルタ処理し、得られた信号902を第二の乗算器706と過大入力推定部102に向けて出力する。
ここで、位相補正部701は、実施例1と同様に全域通過フィルタないしサンプル遅延処理で実現することができるため詳しい説明を省略する。
このように求めた2つの振幅値707、708を制御部105へ向けて出力する。
第一の乗算器705では、入力された信号703とゲイン係数709を乗じて、得られた信号711を加算器713に向けて出力する。
第二の乗算器706では、入力された信号902とゲイン係数710を乗じて、得られた信号712を加算器713に向けて出力する。
加算器713では、入力された2つの信号711、712を加算し、得られた信号を出力信号107として出力する。
実施例1、2では、1つの位相補正部と1つのHPFないしLPFにて周波数特性変形部を実現していたが、これに限らず、複数の位相補正部と複数のHPFないしLPFにて周波数特性変形部を実現しても良い。
図8は、3つの位相補正部と3つHPFにて周波数特性変形部を実現した例を示す図である。以下、本実施例の動作について説明する。
本発明による周波数特性変形装置に入力された入力信号101は3つに分岐され、第一のHPF1101と第二のHPF1102、第三のHPF1103に送られる。
第一のHPF1101では、入力信号をフィルタ処理し、得られた信号1104を第一の位相補正部1107に向けて出力する。
第二のHPF1102では、入力信号をフィルタ処理し、得られた信号1105を第二の位相補正部1108に向けて出力する。
第三のHPF1103では、入力信号をフィルタ処理し、得られた信号1106を第三の位相補正部1109に向けて出力する。
第二の位相補正部1108では、信号の周波数振幅特性を変えずに、第一のHPF1101と第三のHPF1103とを両方処理したときの位相特性と略同一の特性となるように位相特性のみを補正し、得られた信号1111を第二の乗算器1114と過大入力推定部102に向けて出力する。
第三の位相補正部1109では、信号の周波数振幅特性を変えずに、第一のHPF1101と第二のHPF1102とを両方処理したときの位相特性と略同一の特性となるように位相特性のみを補正し、得られた信号1112を第三の乗算器1115と過大入力推定部102に向けて出力する。
本実施例の過大入力推定部102は、スピーカ振動板変位推定部501から構成されている。スピーカ振動板変位推定部501では、ボリューム値や対象スピーカのF0等の情報502を用いて、信号1110を再生したときのスピーカ振動板の変位値を推定し、第一のスピーカ振動板変位値1116を求める。同様に、信号1111を再生したときのスピーカ振動板の変位値を推定し、第二のスピーカ振動板変位値1117を求める。同様に、信号1112を再生したときのスピーカ振動板の変位値を推定し、第三のスピーカ振動板変位値1118を求める。
変位値推定の具体例としては、実施例1と同様の方法で求められるため詳細説明を省略する。
制御部105では、入力された3つのスピーカ振動板変位値1116、1117、1118に対して、夫々異なるゲイン係数を乗じてから加算したときに、振幅値の絶対値が所定の閾値以内に収まるような3つのゲイン係数を求める。ただし、3つのゲイン係数の合計を1とする。
T>ABS(X1×A1+X2×A2+X3×A3) ・・・(2)
A1+A2+A3=1
なお、ABS(x)はxの絶対値を表す。
第一の乗算器1113では、入力された信号1110とゲイン係数1119を乗じて、得られた信号1122を加算器713に向けて出力する。
第二の乗算器1114では、入力された信号1111とゲイン係数1120を乗じて、得られた信号1123を加算器713に向けて出力する。
第三の乗算器1115では、入力された信号1112とゲイン係数1121を乗じて、得られた信号1124を加算器713に向けて出力する。
加算器713では、入力された3つの信号1122、1123、1124を加算し、得られた信号を出力信号107として出力する。
もちろん、位相補正とHPFの個数を増やすことで、さらに通常のHPFに近い特性を実現できるようになる。また、本構成のHPFをLPFに置き換えることにより、高周波数成分の補正が多いような周波数特性補正を施されたデジタル音響信号において、デジタル信号の最大振幅を超えないように信号の周波数特性を変形させることも可能となる。
図10は本発明の別の実施例を示す図である。本実施例では、高域通過フィルタでカットした低域に対し、その高調波を生成、加算する例を示している。以下、本実施例の動作について説明する。
本発明による信号処理装置に入力された入力信号101は分岐され、位相補正部701、HPF702に送られる。
位相補正部701は入力信号の周波数振幅特性を変えずに、HPF702の位相特性と略同一の特性となるように位相特性のみを補正し、得られた信号703を第一の乗算器705、過大入力推定部102と高調波信号生成部2001に向けて出力する。
HPF702は入力信号101をフィルタ処理し、得られた信号704を加算器2003に向けて出力する。
ここで、位相補正部は実施例1と同様に全域通過フィルタないしサンプル遅延処理で実現することができるため詳しい説明を省略する。
低域抽出部2004では、位相補正部701の出力信号703を入力してHPF702でカットされる低域を抽出し、得られた信号2005を高調波生成部2006に向けて出力する。ここで、HPF702でカットされる低域を抽出する低域抽出部2004の実現方法は、図11のように差分器2011から構成して、位相補正部701の出力信号703をHPF702の出力信号704で減算することで実現する方法や、図12のようにLPF2012から構成して、位相補正部701の出力信号703をHPF702のフィルタ仕様と同様なフィルタに通すことで実現する方法がある。
乗算器2008では、高調波生成部2006の出力信号2007を、ユーザ好みのゲイン係数で乗じて、得られた信号2002を加算器2003に向けて出力する。ここで、乗算器2008で乗じるゲイン係数は、固定のゲイン係数を事前に複数用意するなどし、ユーザの好みに応じて変更するものである。
加算器2003では、入力された2つの信号704、2002を加算し、得られた信号2003´を過大入力推定部102と第二の乗算器706に向けて出力する。
変位値推定の具体例としては、実施例1と同様の方法で求められるため詳細説明を省略する。
制御部105では、入力された2つのスピーカ振動板変位値707、708に対して、夫々異なるゲイン係数を乗じてから加算したときに、振幅値の絶対値が所定の閾値以内に収まるような2つのゲイン係数を求める。ただし、2つのゲイン係数の合計を1とする。
このような条件で2つのゲイン係数を変えていくと、異なった低域減衰効果を実現することができる。低域減衰効果の具体例は、実施例1と同様な特性となるため詳細説明を省略する。
図13において、スピーカ振動板変位値707に対するゲイン係数をA1、スピーカ振動板変位値708に対するゲイン係数をA2としたとき、2021はA1=0.0、A2=1.0の特性イメージ、2022はA1=0.1、A2=0.9の特性イメージ、2023は、A1=0.5、A2=0.5の特性イメージを示す。これより、A2の値が大きく、低域の減衰量が大きいほど80Hz以下の低域の高調波が多く加算されることがわかる。
また、ゲイン係数A1、A2の具体的な算出法は、実施例1と同様に求められるため、説明を省略する。
第一の乗算器705では、入力された信号703とゲイン係数709を乗じて、得られた信号711を加算器713に向けて出力する。
第二の乗算器706では、入力された信号2003´とゲイン係数710を乗じて、得られた信号712を加算器713に向けて出力する。
加算器713では、入力された2つの信号711、712を加算し、得られた信号を出力信号107として出力する。
ここでMissing fundamentalとは、2つ以上の周波数の音を聴くと、その差分の周波数の音が聴こえると錯覚する特徴である。
実施例5では、1つの位相補正部と1つのHPFにて周波数特性変形部を実現していたが、これに限らず、複数の位相補正部と複数のHPFにて周波数特性変形部を実現しても良い。
図14は、3つの位相補正部と3つのHPFにて周波数特性変形部を実現した例を示す図である。以下、本実施例の動作について説明する。
本発明による信号処理装置に入力された入力信号101は6つに分岐され、第一のHPF1101、第二のHPF1102、第三のHPF1103と第一のLPF2101と第二のLPF2102、第三のLPF2103に送られる。
第一のHPF1101では、入力信号をフィルタ処理し、得られた信号1104を第一の位相補正部1107に向けて出力する。
第二のHPF1102では、入力信号をフィルタ処理し、得られた信号1105を第二の位相補正部1108に向けて出力する。
第三のHPF1103では、入力信号をフィルタ処理し、得られた信号1106を第三の位相補正部1109に向けて出力する。
第二の位相補正部1108では、信号の周波数振幅特性を変えずに、第一のHPF1101と第三のHPF1103とを両方処理したときの位相特性と略同一の特性となるように位相特性のみを補正し、得られた信号1111を加算器2126に向けて出力する。
第三の位相補正部1109では、信号の周波数振幅特性を変えずに、第一のHPF1101と第二のHPF1102とを両方処理したときの位相特性と略同一の特性となるように位相特性のみを補正し、得られた信号1112を加算器2127に向けて出力する。
ここで、各位相補正部は実施例1と同様に全域通過フィルタないしサンプル遅延処理で実現することができるため詳しい説明を省略する。
第二のLPF2102では、入力信号101を第二のHPF1102と同様なフィルタ仕様を持ったフィルタで処理し、得られた信号2105を第五の位相補正部2108に向けて出力する。
第三のLPF2103では、入力信号101を第三のHPF1103と同様なフィルタ仕様を持ったフィルタで処理し、得られた信号2106を第六の位相補正部2109に向けて出力する。
第五の位相補正部2108では、信号の周波数振幅特性を変えずに、第一のLPF2101と第三のLPF2103とを両方処理したときの位相特性と略同一の特性となるように位相特性のみを補正し、得られた信号2111を第二の高調波生成部2114に向けて出力する。
第六の位相補正部2109では、信号の周波数振幅特性を変えずに、第一のLPF2101と第二のLPF2102とを両方処理したときの位相特性と略同一の特性となるように位相特性のみを補正し、得られた信号2112を第三の高調波生成部2115に向けて出力する。
第二の高調波生成部2114では、信号2111の高調波を生成し、得られた信号2117を乗算器2120に向けて出力する。
第三の高調波生成部2115では、信号2112の高調波を生成し、得られた信号2118を乗算器2121に向けて出力する。
乗算器2119では、信号2116をユーザの好みのゲイン係数で乗じて、得られた信号2122を加算器2125に向けて出力する。
乗算器2120では、信号2117をユーザの好みのゲイン係数で乗じて、得られた信号2123を加算器2126に向けて出力する。
乗算器2121では、信号2118をユーザの好みのゲイン係数で乗じて、得られた信号2124を加算器2127に向けて出力する。
ここで、乗算器2119、2120、2121で乗じるゲイン係数は、固定のゲイン係数を事前に複数用意するなどし、ユーザの好みに応じて変更するものである。
加算器2126では、入力された2つの信号1111、2123を加算し、得られた信号2129を過大入力推定部102と第二の乗算器1114に向けて出力する。
加算器2127では、入力された2つの信号1112、2124を加算し、得られた信号2130を過大入力推定部102と第三の乗算器1115に向けて出力する。
変位値推定の具体例としては、実施例1と同様の方法で求められるため詳細説明を省略する。
制御部105では、入力された3つのスピーカ振動板変位値1116、1117、1118に対して、夫々異なるゲイン係数を乗じてから加算したときに、振幅値の絶対値が所定の閾値以内に収まるような3つのゲイン係数を求める。ただし、3つのゲイン係数の合計を1とする。
このような条件で3つのゲイン係数を変えていくと、異なった低域減衰効果を実現することができる。低域減衰効果の具体例は、実施例3と同様な特性となるため詳細説明を省略する。
また、ゲイン係数A1、A2、A3の具体的な算出法は、実施例3と同様に求められるため、説明を省略する。
第一の乗算器1113では、入力された信号2128とゲイン係数1119を乗じて、得られた信号1122を加算器713に向けて出力する。
第二の乗算器1114では、入力された信号2129とゲイン係数1120を乗じて、得られた信号1123を加算器713に向けて出力する。
第三の乗算器1115では、入力された信号2130とゲイン係数1121を乗じて、得られた信号1124を加算器713に向けて出力する。
加算器713では、入力された3つの信号1122、1123、1124を加算し、得られた信号を出力信号107として出力する。
実施例4、5では、高調波信号生成部2001の出力信号2002をHPFの出力信号と加算する構成であったが、高調波信号生成部2001の出力信号2002を周波数特性変形部103の後段で加算する構成にしても良い。
図16は、実施例4において、高調波信号生成部2001の出力信号2002の加算位置を周波数特性変形部103の後段に変更した構成の実施例を示す図である。以下、本実施例の動作について説明する。
本発明による信号処理装置に入力された入力信号101は分岐され、位相補正部701、HPF702に送られる。
位相補正部701は入力信号の周波数振幅特性を変えずに、HPF702の位相特性と略同一の特性となるように位相特性のみを補正し、得られた信号703を第一の乗算器705、過大入力推定部102と高調波信号生成部2001に向けて出力する。
HPF702は入力信号101をフィルタ処理し、得られた信号704を乗算器706と過大入力推定部102に向けて出力する。
ここで、位相補正部は実施例1と同様に全域通過フィルタないしサンプル遅延処理で実現することができるため詳しい説明を省略する。
変位値推定の具体例としては、実施例1と同様の方法で求められるため詳細説明を省略する。
制御部105では、入力された2つのスピーカ振動板変位値707、708に対して、夫々異なるゲイン係数を乗じてから加算したときに、振幅値の絶対値が所定の閾値以内に収まるような2つのゲイン係数を求める。ただし、2つのゲイン係数の合計を1とする。
このような条件で2つのゲイン係数を変えていくと、異なった低域減衰効果を実現することができる。低域減衰効果の具体例は、実施例1と同様な特性となるため詳細説明を省略する。
また、図16において、スピーカ振動板変位値707に対するゲイン係数をA1、スピーカ振動板変位値708に対するゲイン係数をA2としたとき、ゲイン係数A1、A2の具体的な算出法は、実施例1と同様に求められるため、説明を省略する。
第一の乗算器705では、入力された信号703とゲイン係数709を乗じて、得られた信号711を加算器713に向けて出力する。
第二の乗算器706では、入力された信号704とゲイン係数710を乗じて、得られた信号712を加算器713に向けて出力する。
加算器713では、入力された2つの信号711、712を加算し、得られた信号2203を加算2204に向けて出力する。
低域抽出部2004では、位相補正部701の出力信号を入力し、HPF702でカットされる低域を抽出し、得られた信号2005を高調波生成部2006に向けて出力する。ここで、HPF702でカットされる低域を抽出する低域抽出部2004の実現方法は、実施例4と同様に求められるため、説明を省略する。
乗算器2201では、入力された信号2002とゲイン係数710を乗じて、得られた信号2202を加算器2204に向けて出力する。ここで、乗算器2201では、低域減衰効果に応じて、カットした低域の高調波の加算量を変化させることができる。低域減衰効果に応じた高調波の加算イメージについては、実施例4と同様となるため、説明を省略する。
加算器2204では、入力された2つの信号2202、2203を加算し、得られた信号を出力信号107として出力する。
Claims (11)
- 対象とする信号の周波数特性を変形させるフィルタと、
前記対象とする信号の位相特性を補正し、前記フィルタの位相特性と略同一とする位相補正部と、
前記位相補正部から出力された信号のゲインを調整する第一の乗算器と、
前記フィルタから出力された信号のゲインを調整する第二の乗算器と、
前記第一の乗算器のゲイン係数と第二の乗算器のゲイン係数の合計が一定値となるように前記第一および第二の乗算器のゲイン係数を決定する係数決定部と、
前記第一の乗算器および第二の乗算器から出力される2つの信号を加算する加算器とを備えたことを特徴とする周波数特性変形装置。 - 前記フィルタは、高域通過フィルタで構成され、
目標とする特性のカットオフ周波数を高くする場合には、前記係数決定部が前記第二の乗算器のゲイン係数を最大値に近づけ、目標とする特性のカットオフ周波数を低くする場合には、前記係数決定部が前記第二の乗算器のゲイン係数を0に近づけることにより目標とする特性のカットオフ周波数を変化させることを特徴とする請求項1記載の周波数特性変形装置。 - 前記フィルタは、低域通過フィルタで構成され、
目標とする特性のカットオフ周波数を高くする場合には、前記係数決定部が前記第二の乗算器のゲイン係数を0に近づけ、目標とする特性のカットオフ周波数を低くする場合には、前記係数決定部が前記第二の乗算器のゲイン係数を最大値に近づけることにより目標とする特性のカットオフ周波数を変化させることを特徴とする請求項1記載の周波数特性変形装置。 - カットオフ周波数の異なる複数の高域通過フィルタと、前記対象とする信号の位相特性を補正し、前記各高域通過フィルタの位相特性と略同一とする位相補正部と、前記高域通過フィルタおよび前記位相補正部から出力された信号のゲインを調整する複数の乗算器と、前記複数の乗算器のゲイン係数の合計が一定値となるように各ゲイン係数を決定する係数決定部とを備え、
前記対象とする信号を前記複数の高域通過フィルタに通して複数のフィルタ出力信号を生成し、生成された各フィルタ出力信号の位相特性を、他の高域通過フィルタの位相特性に相当する位相補正部を用いて補正して各フィルタ出力信号の位相特性を略同一とし、目標とする特性のカットオフ周波数を低くする場合には、前記係数決定部にてカットオフ周波数の低いフィルタからの出力信号に対応する乗算器のゲイン係数を最大値に近づけ、目標とする特性のカットオフ周波数を高くする場合には、前記係数決定部にてカットオフ周波数の高いフィルタからの出力信号に対応する乗算器のゲイン係数を最大値に近づけ、前記位相補正された各フィルタ出力信号に対して、前記係数決定部が決定した各ゲイン係数により重みづけを行った後、前記加算器によって各信号を加算することで、目標とする特性のカットオフ周波数を変化させることを特徴とする請求項2記載の周波数特性変形装置。 - カットオフ周波数の異なる複数の低域通過フィルタと、前記対象とする信号の位相特性を補正し、前記各低域通過フィルタの位相特性と略同一とする位相補正部と、前記低域通過フィルタおよび前記位相補正部から出力された信号のゲインを調整する複数の乗算器と、前記複数の乗算器のゲイン係数の合計が一定値となるように各ゲイン係数を決定する係数決定部とを備え、
前記対象とする信号を前記複数の低域通過フィルタに通して複数のフィルタ出力信号を生成し、生成された各フィルタ出力信号の位相特性を、他の低域通過フィルタの位相特性に相当する位相補正部を用いて補正して各フィルタ出力信号の位相特性を略同一とし、目標とする特性のカットオフ周波数を低くする場合には、前記係数決定部にてカットオフ周波数の低いフィルタからの出力信号に対応する乗算器のゲイン係数を最大値に近づけ、目標とする特性のカットオフ周波数を高くする場合には、前記係数決定部にてカットオフ周波数の高いフィルタからの出力信号に対応する乗算器のゲイン係数を最大値に近づけ、前記位相補正された各フィルタ出力信号に対して、前記係数決定部が決定した各ゲイン係数により重みづけを行った後、前記加算器によって各信号を加算することで、目標とする特性のカットオフ周波数を変化させることを特徴とする請求項3記載の周波数特性変形装置。 - 前記高域通過フィルタを通してカットされる低域を抽出する低域抽出部と、
前記低域抽出部から出力された信号の高調波を生成する高調波生成部とを備える
ことを特徴とする請求項2記載の周波数特性変形装置。 - 前記高域通過フィルタを通してカットされる低域を抽出する低域抽出部と、
前記低域抽出部から出力された信号の高調波を生成する高調波生成部とを備える
ことを特徴とする請求項4記載の周波数特性変形装置。 - 前記高域通過フィルタを通してカットされる低域を抽出する低域抽出部と、
前記低域抽出部から出力された信号の高調波を生成する高調波生成部と、
前記高調波生成部から出力された信号と前記高域通過フィルタから出力された信号を加算する加算器とを備え、
前記加算器から出力された信号のゲインを前記第二の乗算器で変化させ、前記高調波生成部から出力された信号のゲインを低域の減衰特性に応じて変化させることを特徴とする請求項2記載の周波数特性変形装置。 - 前記高域通過フィルタを通してカットされる低域を抽出する低域抽出部と、
前記低域抽出部から出力された信号の高調波を生成する高調波生成部と、
前記第二の乗算器で乗じるゲイン係数を、前記高調波生成部から出力される信号に乗じる乗算器とを備え、
前記高調波生成部から出力された信号のゲインを低域の減衰特性に応じて変化させることを特徴とする請求項2記載の周波数特性変形装置。 - 前記高域通過フィルタを通してカットされた低域を抽出する複数の低域抽出部と、
前記複数の低域抽出部から出力された信号の高調波を生成する複数の高調波生成部と、
前記高調波生成部から出力された複数の信号と前記高域通過フィルタから出力された複数の信号を加算する複数の加算器とを備え、
前記加算器から出力された複数の信号のゲインを請求項4記載の前記複数の乗算器で変化させ、前記高調波生成部から出力された複数の信号のゲインを低域の減衰特性に応じて変化させることを特徴とする請求項4記載の周波数特性変形装置。 - 前記高域通過フィルタを通してカットされた低域を抽出する複数の低域抽出部と、
前記複数の低域抽出部から出力された信号の高調波を生成する複数の高調波生成部と、
請求項4記載の前記乗算器で乗じる複数のゲイン係数を、前記複数の高調波生成部から出力される信号に乗じる複数の乗算器とを備え、
前記高調波生成部から出力された複数の信号のゲインを低域の減衰特性に応じて変化させることを特徴とする請求項4記載の周波数特性変形装置。
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- 2012-12-14 US US14/373,721 patent/US9552826B2/en active Active
- 2012-12-14 WO PCT/JP2012/082546 patent/WO2013183185A1/ja active Application Filing
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WO2018167834A1 (ja) * | 2017-03-14 | 2018-09-20 | 三菱電機株式会社 | 音響信号処理装置 |
US10771895B2 (en) | 2017-03-14 | 2020-09-08 | Mitsubishi Electric Corporation | Audio signal processing device |
Also Published As
Publication number | Publication date |
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CN104137568A (zh) | 2014-11-05 |
CN104137568B (zh) | 2017-04-12 |
US20150030181A1 (en) | 2015-01-29 |
US9552826B2 (en) | 2017-01-24 |
DE112012006457B4 (de) | 2020-04-02 |
DE112012006457T5 (de) | 2015-02-19 |
WO2013183103A1 (ja) | 2013-12-12 |
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