The present invention relates to an acoustic processing device and an acoustic processing method.
In a listening environment such as a vehicle interior, for example, standing waves may interfere with each other to cause a dip in a frequency region, thereby deteriorating sound quality or reducing sound pressure.
In order to suppress such a dip, for example, an acoustic processing device that parametrically controls an amplitude characteristic in a frequency domain is known. As an example, an acoustic processing device described in Patent Document 1 reads a filter coefficient group from a memory and applies the filter coefficient group to a filter. The amplitude characteristic is adjusted and the dip is suppressed by the filter processing by the filter. Patent Document 1: Japanese Unexamined Patent Application 2013-219731.
In a conventional acoustic processing device including Patent Document 1, a filter coefficient group calculated under a predetermined condition (for example, under a condition that only a gain is changed by a predetermined value and other parameters are fixed) is stored in advance in a memory. In other words, only a limited filter coefficient group is stored in the memory. Therefore, the degree of freedom of the filter is low.
In order to improve the degree of freedom of the filter, a method of calculating the filter coefficient group under the condition that each parameter is finely changed may be considered. However, in this case, it is necessary to calculate an enormous number of combinations of parameters. Since an enormous amount of time is required for the calculation processing, it is not easy to adopt such a method.
Therefore, in light of the foregoing, an object of the present application is to provide an acoustic processing device and acoustic processing method that can improve the degree of freedom of the filter.
The acoustic processing device unit according to an embodiment of the present application is provided with: an input reception unit configured to receive an input of a value of a parameter defining a phase difference to be provided between a pair of audio signals; a filter coefficient calculation unit configured to calculate a filter coefficient group corresponding to each of the pair of audio signals based on the input value received by the input reception unit; and a filter processing unit configured to provide a phase difference defined by the input value between the pair of audio signals by performing filter processing on each of the pair of audio signals based on the filter coefficient group calculated by the filter coefficient calculation unit. The parameter above includes a center frequency of the phase difference provided between the pair of audio signals, a quality factor, and a phase difference at the center frequency.
According to an embodiment of the present application, an acoustic processing device and an acoustic processing method are provided wherein it is possible to improve the degree of freedom of a filter.
The following description relates to an acoustic processing device and acoustic processing method according to an embodiment of the present application.
The acoustic processing device 2 is an example of a computer, and performs phase control by an Infinite Impulse Response (IIR) all-pass filter pair for a pair of transducers to reduce phase interference of a sound field. For example, the acoustic processing device 2 suppresses the occurrence of a dip in a frequency domain due to the interference of standing waves, thereby suppressing the deterioration of sound quality and the reduction of sound pressure.
The speakers SPFR and SPFL are an example of a pair of transducers. The speaker SPFR is a right front speaker embedded in a right door unit (driver's seat side door unit). The speaker SPFL is a left front speaker embedded in a left door unit (passenger's seat side left door unit).
The vehicle A may have yet another speaker (for example, rear speaker) installed (that is, three or more speakers installed). In this case, the transducer pair to be processed is not limited to the speaker SPFR and the speaker SPFL. The transducer pair to be processed may be, for example, two rear speakers installed at left and right positions of a rear seat. Furthermore, the transducer pair to be processed may be the speaker SPFR and any one of the rear speakers, or may be the speaker SPFL and any one of the rear speakers.
The measuring device 3 is a well-known device that measures a frequency characteristic of sound, and is installed at a predetermined listening point (a driver's seat, a front passenger's seat, a rear seat, or the like). The measuring device 3 collects sounds output from the speakers SPFR and SPFL using a microphone, analyzes a frequency characteristic of the collected sounds, and displays the analyzed frequency characteristic on a display unit.
An operator (user) confirms the frequency characteristic displayed on the display unit of the measuring device 3. For example, when a dip is confirmed, the operator inputs, into the acoustic processing device 2, a parameter value suitable for reducing the phase interference of the sound field at the listening point. Here, the parameters are parameters for calculating a filter coefficient group to be applied to the all-pass filter pair, and specifically, are the center frequency of the phase difference between the pair of audio signals, the quality factor, and the phase difference at the center frequency.
Hereinafter, the parameters of the center frequency, the quality factor (also referred to as Q-value), and the phase difference at the center frequency input by the operator are referred to as a center frequency parameter ω, a quality factor parameter θ, and a phase difference parameter φ, respectively.
The measuring device 3 may not be installed at the listening point. In this case, the operator sits at the listening point and inputs a parameter value suitable for reducing the phase interference of the sound field at the listening point into the acoustic processing device 2 while checking the sound actually heard. That is, the operator may adjust the sound field at the listening point depending on his/her own auditory sense.
Hereinafter, the audio signal of the R channel corresponding to the speaker SPFR will be referred to as audio signal SR. Also, the L channel audio signal corresponding to the speaker SPFL will be referred to as an audio signal SL.
A phase difference is provided between a pair of audio signals (between the audio signal SR and the audio signal SL) by the all-pass filter pair to which the filter coefficient group is applied. Hereinafter, the phase difference between the audio signal SR and the audio signal SL is referred to as a phase difference PH in order to be distinguished from the phase difference (phase difference parameter φ) at the center frequency.
In the present embodiment, the operator can provide an appropriate phase difference PH between the audio signal SR and the audio signal SL by inputting an appropriate parameter value into the acoustic processing device 2, thereby suppressing occurrence of a dip in a frequency domain due to the interference of standing waves and suppressing the deterioration of sound quality and the reduction in sound pressure.
In
Here, in the conventional configuration, only the filter coefficient group stored in advance in the memory can be used to execute the filter processing. In this configuration, since only a limited filter coefficient group can be used, it is difficult to realize the adjustment of the phase difference PH by the all-pass filter pair with high resolution. Further, it is necessary to secure a storage area for storing a large number of filter coefficient groups in the memory.
On the other hand, the acoustic processing device 2 is configured to apply the phase difference PH input by the operator (for example, the phase difference defined by the values of the center frequency parameter ω, the quality factor parameter θ, and the phase difference parameter φ illustrated in
In the acoustic processing device 2, since the filter can be set based on an arbitrary parameter value, the degree of freedom of the filter is higher than that of the conventional configuration. Further, it is not necessary to secure a storage area for storing a large number of filter coefficient groups in the memory.
The player 10 is connected to a sound source. The player 10 plays an audio signal input from the sound source, which is then output to the LSI 11.
Examples of the sound source include disc media such as CDs (Compact Discs), SACDs (Super Audio CDs), and the like, and storage media such as HDDs (Hard Disk Drive), USBs (Universal Serial Bus), and the like that store digital audio data. A telephone (for example, a feature phone or smartphone) may be the sound source. In this case, the player 10 outputs a voice signal during a call input from the telephone through to the LSI 11.
The LSI 11 is provided with a CPU (Central Processing Unit), RAM (Random Access Memory), ROM (Read Only Memory), and the like. The CPU of the LSI 11 includes a single processor or a multiprocessor (in other words, at least one processor) that executes a program written in the ROM of the LSI 11, and comprehensively controls the acoustic processing device 2.
The LSI 11 executes a program developed in a work area such as a RAM to receive input of a value of a parameter (specifically, the center frequency parameter ω, the quality factor parameter θ, and the phase difference parameter φ) defining the phase difference PH to be provided between the pair of audio signals, calculates the filter coefficient group corresponding to each of the pair of audio signals based on the received input value, and perform filter processing on each of the pair of audio signals based on the calculated filter coefficient group to provide the phase difference PH defined by the input value between the pair of audio signals.
By executing the program, for example, a suitable phase difference PH is provided between the audio signal SR and the audio signal SL, occurrence of a dip in a frequency domain due to the interference of standing waves is suppressed, and deterioration in sound quality and reduction in sound pressure are suppressed.
The audio signals SR and SL after the filter processing by the LSI 11 are converted to an analog signal by the D/A converter 12. The analog signal is amplified by the amplifier 13 and output to the speakers SPFR and SPFL. As a result, music recorded in the sound source, for example, is reproduced in the vehicle interior from the speakers SPFR and SPFL.
In the present embodiment, a vehicle-mounted acoustic processing system 1 is exemplified. However, even in a listening environment such as a room of a building, a dip may occur in a frequency domain due to interference of standing waves at a listening point. Therefore, the acoustic processing system 1 may be implemented for listening environments other than a vehicle interior.
The display unit 14 is a device that displays various screens, such as a settings screen, and examples include displays such as LCDs (Liquid Crystal Displays) and organic EIs (Electro Luminescence). The display unit 14 may be configured to include a touch panel.
The operation unit 15 includes operators such as switches, buttons, knobs, and wheels, that are mechanical systems, capacitance non-contact systems, membrane systems, or the like. When the display unit 14 includes a touch panel, the touch panel also forms a portion of the operation unit 15.
For example, when the operator inputs values of the center frequency parameter ω, the quality factor parameter θ, and the phase difference parameter φ using the operation unit 15, a phase difference PH (see
As illustrated in
The input reception unit 210 receives an input by an operator using the operation unit 15, which is an input of a value of a parameter defining a phase difference to be provided between a pair of audio signals.
The filter coefficient calculation unit 220 calculates a filter coefficient group corresponding to each of the pair of audio signals based on the input value received by the input reception unit 210.
Specifically, the filter coefficient calculation unit 220 calculates a filter coefficient group FCR (an example of a first filter coefficient group) corresponding to the R channel and a filter coefficient group FCL (an example of a second filter coefficient group) corresponding to the L channel based on the values of the center frequency parameter ω, the quality factor parameter θ, and the phase difference parameter φ input using the operation unit 15.
The filter processing unit 230 performs filter processing on each of the pair of audio signals based on the filter coefficient group calculated by the filter coefficient calculation unit 220, thereby providing the phase difference PH defined by the input value between the pair of audio signals.
The R filter unit 232 is an example of a first all-pass filter, and the filter coefficient group FCR is applied thereto. The L filter unit 234 is an example of a second all-pass filter, and the filter coefficient group FCL is applied thereto. The R filter unit 232 and the L filter unit 234 constitute, for example, a direct form I IIR all-pass filter.
To the R filter unit 232, the audio signal SR of the R channel is input from the player 10 and the filter coefficient group FCR is input from the filter coefficient calculation unit 220. The R filter unit 232 performs filter processing on the audio signal SR using the filter coefficient group FCR.
To the L filter unit 234, the audio signal SL of the L channel is input from the player 10 and the filter coefficient group FCL is input from the filter coefficient calculation unit 220. The L filter unit 234 performs filter processing on the audio signal SL using the filter coefficient group FCL.
As a result of the filtering processing performed by the R filter unit 232 and the L filter unit 234, the phase difference PH (the phase difference defined by the values of the center frequency parameter ω, the quality factor parameter θ, and the phase difference parameter φ) specified by the operator is provided given to the audio signal pair. The audio signals SR and SL to which the phase difference PH is provided are output as sound into the vehicle interior by the speakers SPFR and SPFL via the D/A converter 12 and the amplifier 13. By providing a suitable phase difference PH in the R filter unit 232 and the L filter unit 234, occurrence of a dip in a frequency domain due to the interference of standing waves is suppressed, and deterioration of sound quality and reduction in sound pressure are suppressed.
A method of calculating a filter coefficient group by the filter coefficient calculation unit 220 will be specifically described.
Characteristics of the R filter unit 232 and the L filter unit 234 are expressed as in Equation 1, using a biquad transfer function H(z) of Z transform. In Equation 1, a0, a1, and a2 represent filter coefficients of the denominator of the biquad transfer function H(z). b0, b1, and b2 represent filter coefficients of the numerator of the biquad transfer function H(z). The filter coefficient calculation unit 220 calculates a total of the above six filter coefficients for each of the R channel and the L channel.
Equation 2 is derived when normalization is performed so that the filter coefficient a0 becomes 1 in Equation 1. Further, by transforming Equation 2, the characteristics of the R filter unit 232 and the L filter unit 234 implemented as direct form I can be expressed by Equation 3. In Equation 3, y represents the output signal and x represents the input signal.
The R filter unit 232 and the L filter unit 234, which are all-pass filters, can be represented as analog prototype filters having a transfer function H(s) as in Equation 4.
A filter coefficient group (six filter coefficients a0, a1, a2, b0, b1, and b2) shown in the following Equation 5 is derived from Equations 3 and 4. In order to derive Equation 5, for example, a known bilinear transform is used, but the method of deriving Equation 5 is not limited thereto.
b
0=1−α
b
1=−2 cos ω0
b
2=1+α
a
0=1+α
a
1=−2 cos ω0
a
2=1−α (Equation 5)
Further, intermediate variables ω0 and a included in Equation 5 are represented by Equations 6 and 7, respectively. In Equation 6, f0 represents the center frequency of the filter calculated by the filter coefficient calculation unit 220. In Equation 7, β represents the bandwidth of the filter calculated by the filter coefficient calculation unit 220.
The filter coefficient calculation unit 220 calculates band edge frequencies fC+ and fC− corresponding to the R channel and the L channel, respectively (step S101).
The filter coefficient calculation unit 220 obtains a bandwidth 13 based on the band edge frequencies fC+ and fC− calculated in step S101 (step S102).
The filter coefficient calculation unit 220 further substitutes the band edge frequency fC+ as the center frequency f0 into Equation 6 to calculate the intermediate variable ω0 corresponding to the R channel, and substitutes the band edge frequency fC− as the center frequency f0 into Equation 6 to calculate the intermediate variable ω0 corresponding to the L channel (step S103).
The filter coefficient calculation unit 220 substitutes the bandwidth 13 and the intermediate variable ω0 corresponding to the R channel into Equation 7 to calculate the intermediate variable α corresponding to the R channel, and substitutes the bandwidth β and the intermediate variable ω0 corresponding to the L channel into Equation 7 to calculate the intermediate variable α corresponding to the L channel (step S104).
The filter coefficient calculation unit 220 substitutes the intermediate variables ω0 and a corresponding to the R channel into Equation 5 to obtain the filter coefficient group FCR, and substitutes the intermediate variables ω0 and a corresponding to the L channel into Equation 5 to obtain the filter coefficient group FCL (step S105).
Next, the filter coefficient calculation processing illustrated in
In step S101, the band edge frequencies fC+ and fC− are obtained by using the functions f1(ω, θ, φ) and f2(ω, θ, φ) with the center frequency parameter ω, the quality factor parameter θ and the phase difference parameter φ as variables, respectively.
The distance D on the frequency axis (logarithmic axis) between the band edge frequencies fC+ and fC− calculated by the function f1(ω, θ, φ) and the function f2(ω, θ, φ) has a negative correlation with the quality factor parameter θ and a positive correlation with the phase difference parameter φ. According to the function f1(ω, θ, φ) and the function f2(ω, θ, φ), the band edge frequencies fC+ and fC− are calculated, being symmetrical centered on a center frequency (a value of the center frequency parameter ω, hereinafter given the reference numeral fC) on a frequency axis (for example, a logarithmic axis) in consideration of human auditory characteristics.
The characteristics of the function f1(ω, θ, φ) and the function f2(ω, θ, φ) will be described with reference to
Drawings labeled “Linear diff” show the distance D on the linear axis. Drawings labeled “Log diff” show the distance D on the logarithmic axis. Drawings labeled “Linear plot” show the horizontal axis as a linear axis. Drawings labeled “log-log plot” show a log-log graph (log-log axis).
As illustrated in
In step S101, band edge frequencies fC+ and fC− are calculated using the function f1(ω, θ, φ) and the function f2(ω, θ, φ). The distance D on the frequency axis between the band edge frequencies fC+ and fC− calculated by these functions has a negative correlation with the quality factor parameter θ and a positive correlation with the phase difference parameter φ. Furthermore, the band edge frequencies fC+ and fC− calculated by these functions appear at positions symmetrical centered on the center frequency fC on the logarithmic axis in which the human auditory characteristics are taken into consideration.
In this way, the filter coefficient calculation unit 220 calculates the band edge frequency fC+ (an example of the frequency of the first control target for one of the pair of audio signals) and the band edge frequency fC− (an example of the frequency of the second control target for the other of the pair of audio signals) based on the input values (the values of the center frequency parameter ω, the quality factor parameter θ, and the phase difference parameter φ).
In addition, the filter coefficient calculation unit 220 gives the input values to the function f1(ω, θ, φ) (an example of a first function) and the function f2(ω, θ, φ) (an example of a second function) to obtain one of a pair of frequencies (for example, the band-end frequency fC+) located symmetrically around the center frequency on the logarithmic axis as the frequency of the first control object and the other of the pair of frequencies (for example, the band edge frequency fC−) as the frequency of the second control object. The distance D, which is the difference value between the function f1(ω, θ, φ) and the function f2(ω, θ, φ), has a negative correlation with the quality factor parameter θ and has a positive correlation with the phase difference parameter φ.
For example, a constant term (numerically, for example, a value in consideration of a ⅓ octave band) in consideration of human auditory characteristics may be incorporated into the function f1(ω, θ, φ) and the function f2(ω, θ, φ).
Furthermore, a parametric equalizer is widely known as a technique for adjusting amplitude. In the parametric equalizer, for example, a center frequency, a bandwidth, and a gain are adopted as input parameters. Therefore, an adjustment term for approximating the curve of the peak determined by the input values of the center frequency parameter ω, the quality factor parameter θ, and the phase difference parameter φ (see
In step S102, the bandwidth β is determined by Equation 8.
The characteristics of Expression 8 will be described using the
In these drawings, the fixed value of the center frequency parameter ω is 1000 (kHz), the fixed value of the quality factor parameter θ is 1, and the fixed value of the phase difference parameter φ is 180°.
As illustrated in
As illustrated in each of
As illustrated in each of
For convenience, the value of the function β1 when the phase difference parameter φ is 0° is assumed to be 180. In this case, the function β1 can be expressed as, for example, “a function having a negative correlation with the phase difference parameter φ, and a function of the phase difference parameter φ having a predetermined closed interval ([0,180] in the present embodiment) as a domain and a value range.”
As a variation, the function β2 can also be represented by Equation 9 or Equation 10.
According to
As described above, in step S102, the filter coefficient calculation unit 220 calculates the bandwidth β of the filter based on the frequencies of the first control target and the frequencies of the second control target (that is, the band edge frequencies fC+ and fC−). In addition, the filter coefficient calculation unit 220 calculates the reference term β0 based on the difference absolute value between the frequencies of the first and second control objects and the center frequency parameter ω, and obtains the bandwidth β by multiplying the calculated reference term β0 by a function β1 (an example of a third function) having a negative correlation with the phase difference parameter φ and having a predetermined closed interval as a domain and a value range, and a function β2 (an example of a fourth function) having a negative correlation with the phase difference parameter φ and converging from 2 to √2 as the phase difference parameter φ increases.
In step S103, the intermediate variable ω0 corresponding to the R channel is obtained by Equation 11, and the intermediate variable ω0 corresponding to the L channel is obtained by Equation 12.
ω0=2πfC+ (Equation 11)
ω0=2πfC− (Equation 12)
In step S104, the intermediate variable αfC+ corresponding to the R channel is obtained by Equation 13, and the intermediate variable αfC− corresponding to the L channel is obtained by Equation 14. In Equation 13, the notation of “fC+” is exactly ω0 corresponding to the R channel, that is, “2πfC+.” In addition, in Equation 14, the notation of “fC−” is exactly ω0 corresponding to the L channel, that is, “2πfC−.”
In step S105, the filter coefficient group FCR is obtained by substituting the intermediate variable ω0 obtained by Equation 11 and the intermediate variable αfC+ obtained by Equation 13 into Equation 5. Further, the filter coefficient group FCL is obtained by substituting the intermediate variable ω0 obtained by Equation 12 and the intermediate variable αfC− obtained by Equation 14 into Equation 5.
As described above, in steps S103 to S105, the filter coefficient calculation unit 220 calculates the filter coefficient group FCR (an example of the first filter coefficient group corresponding to one audio signal) based on the frequency and bandwidth of the first control target (that is, the band edge frequency fC+ and the bandwidth β), and calculates the filter coefficient group FCL (an example of the second filter coefficient group corresponding to the other audio signal) based on the frequency and bandwidth of the second control target (that is, the band edge frequency fC− and the bandwidth β).
By the all-pass filter pair (the R filter unit 232 and the L filter unit 234) to which such a filter coefficient group is provided, the phase difference PH that satisfies the values of the center frequency parameter ω, the quality factor parameter θ, and the phase difference parameter φ designated by the operator is provided to the audio signal SR and the audio signal SL.
According to the present embodiment, since the filter can be set based on arbitrary values of the center frequency parameter ω, the quality factor parameter θ, and the phase difference parameter φ, the degree of freedom of the filter is higher than that of the conventional configuration. In other words, since the parameter values can be arbitrarily determined, the resolution of the phase difference that can be set for the audio signal SR and the audio signal SL is high. Further, by inputting a parameter value, a phase difference corresponding to the parameter value can be immediately provided to the audio signal SR and the audio signal SL. Further, it is not necessary to secure a storage area for storing a large number of filter coefficients in the memory.
The above is a description of an exemplary embodiment of the present application. The embodiments of the present application are not limited to those described above, and various modifications are possible within the scope of the technical concept of the present invention. For example, appropriate combinations of embodiments and the like that are explicitly indicated by way of example in the specification or obvious embodiments and the like are also included in the embodiments of the present application.
For example, in the above description, an acoustic processing device 2 was shown that is provided with only one pair of all-pass filters, but a pair of all-pass filters is required for each band to which a phase difference is to be applied, for example. Therefore, the acoustic processing device 2 may be configured to include a plurality of pairs of all-pass filters so as to be able to apply phase differences to a plurality of bands.
In addition, in the above-described embodiment, the filter coefficient calculation unit 220 performs calculation using, for example, a logarithm, but may perform an equivalent calculation without using a logarithm.
In addition, in the above-described embodiment, the operator inputs each parameter value to the acoustic processing device 2. However, in another embodiment, for example, a higher-level program different from the program installed in the acoustic processing device 2 may automatically provide each parameter value to the acoustic processing device 2.
As an example, a device in which a higher program is installed acquires an image captured by an in-vehicle camera, extracts a person from the acquired captured image, and acquires a seating position of each extracted person. This device provides each parameter value to the acoustic processing device 2 for a transducer pair according to the acquired seating position. For example, when only a person seated in the driver's seat is extracted, the device provides each parameter value for the speakers SPFR and SPFL to the acoustic processing device 2. In addition, for example, when a person seated on the driver's seat and a person seated on the rear seat are extracted, the device provides each parameter value for the speakers SPFR and SPFL and two rear speakers (that is, each of two transducer pairs) to the acoustic processing device 2.
Number | Date | Country | Kind |
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2022-137368 | Aug 2022 | JP | national |