WO2019208285A1 - 音像再現装置、音像再現方法及び音像再現プログラム - Google Patents
音像再現装置、音像再現方法及び音像再現プログラム Download PDFInfo
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
- H04S7/30—Control circuits for electronic adaptation of the sound field
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
- G10K11/00—Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
- G10K11/18—Methods or devices for transmitting, conducting or directing sound
- G10K11/26—Sound-focusing or directing, e.g. scanning
- G10K11/34—Sound-focusing or directing, e.g. scanning using electrical steering of transducer arrays, e.g. beam steering
- G10K11/341—Circuits therefor
- G10K11/346—Circuits therefor using phase variation
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
- G10K11/00—Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
- G10K11/18—Methods or devices for transmitting, conducting or directing sound
- G10K11/26—Sound-focusing or directing, e.g. scanning
- G10K11/34—Sound-focusing or directing, e.g. scanning using electrical steering of transducer arrays, e.g. beam steering
- G10K11/341—Circuits therefor
- G10K11/348—Circuits therefor using amplitude variation
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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
- H04R3/12—Circuits for transducers, loudspeakers or microphones for distributing signals to two or more loudspeakers
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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
- H04R5/00—Stereophonic arrangements
- H04R5/02—Spatial or constructional arrangements of loudspeakers
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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
- H04R5/00—Stereophonic arrangements
- H04R5/04—Circuit arrangements, e.g. for selective connection of amplifier inputs/outputs to loudspeakers, for loudspeaker detection, or for adaptation of settings to personal preferences or hearing impairments
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S3/00—Systems employing more than two channels, e.g. quadraphonic
- H04S3/008—Systems employing more than two channels, e.g. quadraphonic in which the audio signals are in digital form, i.e. employing more than two discrete digital channels
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S2400/00—Details of stereophonic systems covered by H04S but not provided for in its groups
- H04S2400/11—Positioning of individual sound objects, e.g. moving airplane, within a sound field
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S2420/00—Techniques used stereophonic systems covered by H04S but not provided for in its groups
- H04S2420/13—Application of wave-field synthesis in stereophonic audio systems
Definitions
- the present invention relates to a sound image reproduction technique for creating a virtual sound source in a space.
- Patent Document 1 There is a method called wavefront synthesis for sound image reproduction technology for creating a virtual sound source in a screening space (Patent Document 1).
- Patent Document 1 After collecting sound signals at points where sound signals are recorded by microphones installed at a plurality of points, the arrival directions of the sound signals in the vertical and horizontal directions are analyzed, and a plurality of sound signals installed in the presentation space are analyzed. The sound signal of the recording venue is physically reproduced using the speakers.
- Non-Patent Document 1 a drive signal based on the drive function derived from the first type Rayleigh integration is given to the speaker array, so that the virtual sound source is placed in front of the speaker.
- Non-Patent Document 2 a technique using a method of circle harmony expansion is known (Non-Patent Document 2).
- Circle harmonic expansion is a technique for expressing the directivity of sound by expanding acoustic signals observed by a microphone array arranged in a circle around a sound source into a circular harmonic series.
- the sound source with the directional characteristics modeled on the sound collection side is reproduced by using the drive signal of the drive function obtained from the circular harmonic series obtained on the recording side from the speaker array arranged in a circle. Can do.
- Patent Document 1 has high reproducibility in reproducing a virtual sound source in order to faithfully reproduce the acoustic signal at the recording point, but it requires not only a speaker array but also a microphone array, which increases the scale of the entire system. To do.
- the recorded sound is reproduced faithfully, it is difficult to edit content such as adding a sound effect that does not exist every day as a special effect, as represented by a movie, for example.
- the sound signals emitted from a plurality of sound sources are mixed in the microphone at the same time, it is extremely difficult to edit such as taking out each sound source and adjusting the position and sound quality.
- Non-Patent Document 1 does not require a microphone array to generate a virtual sound source, generates sound signals for channels corresponding to a plurality of speakers from a monaural sound source recorded from a normal microphone, A virtual sound source can be created. Since a monaural sound source is used, the scale of the entire system can be reduced, and content editing can be performed easily. However, since omnidirectionality is assumed as the radiation characteristic of the virtual sound source, it is not possible to generate a directional sound source using the virtual sound source.
- the present invention has been made in view of the above circumstances, and can be applied to a monaural sound source, and can provide directivity to a virtual sound source in a space, a sound image reproduction method, a sound image reproduction method, and a sound image reproduction program.
- the purpose is to provide.
- a sound image reproduction device is a sound image reproduction device that generates a virtual sound source in a space using a plurality of speakers arranged in a straight line.
- a focus position determination unit that determines the position of each virtual sound source for generating the sound source, and a driving function for driving a speaker used to generate a virtual sound source at the position of each virtual sound source.
- a filter function determining unit that calculates an impulse response vector for each speaker by performing inverse Fourier transform on the driving function for each speaker to which a weight different from the others is applied, and for each speaker with respect to one input acoustic signal
- a convolution operation unit that outputs the respective acoustic signals to the plurality of speakers, respectively.
- the sound image reproduction device is a sound image reproduction device that generates a virtual sound source in a space using a plurality of speakers arranged in a straight line, and each virtual sound source for generating a plurality of virtual sound sources in a circular shape.
- the focus position determination unit for determining the position of each of the virtual sound sources, and the speaker driving function used for generating the virtual sound sources at the positions of the respective virtual sound sources are given different weights to some of the respective virtual sound sources.
- a filter operation unit that performs inverse Fourier transform on a driving function for each speaker, and convolves an impulse response vector for each speaker calculated in advance with respect to one input acoustic signal and outputs a weighted acoustic signal; For each speaker, the output time of the weighted sound signal is delayed by the time required to travel at the speed of sound through the distance between the speaker and the plurality of virtual sound sources.
- a delay adjustment unit that outputs a delayed acoustic signal, and a gain determined from a distance between each of the speakers and the plurality of virtual sound sources for each speaker, the plurality of virtual sound sources And a gain multiplier for multiplying and outputting each of the delayed acoustic signals.
- the sound image reproduction device is the sound image reproduction device according to claim 1 or 2, wherein the drive function for each speaker circularly develops the directivity characteristics of the virtual sound source in advance with respect to the plurality of virtual sound sources.
- the n-th order circular harmonic series obtained in this way is divided for each order by a two-dimensional green function circularly expanded with respect to the virtual sound source, and the divided values are summed to obtain a weighting coefficient for each virtual sound source, It is a function obtained by weighted averaging the weighting coefficient of each virtual sound source and the driving function for driving the speaker.
- the sound image reproduction method is a sound image reproduction method for generating a virtual sound source in a space using a plurality of speakers arranged in a straight line, and the sound image reproduction device generates a plurality of virtual sound sources in a circle. Determining a position of each virtual sound source and a driving function for driving a speaker used for generating a virtual sound source at the position of each virtual sound source, a part of the virtual sound source having a weight different from others A step of calculating an impulse response vector for each speaker by performing inverse Fourier transform on the given drive function for each speaker, and convolution of the impulse response vector for each speaker with each input acoustic signal, And a step of outputting each of the acoustic signals to the plurality of speakers.
- the sound image reproduction method is a sound image reproduction method for generating a virtual sound source in a space using a plurality of speakers arranged in a straight line, and the sound image reproduction device generates a plurality of virtual sound sources in a circle.
- Determining a position of each virtual sound source and a driving function for driving a speaker used for generating a virtual sound source at the position of each virtual sound source, a part of the virtual sound source having a weight different from others By applying inverse Fourier transform to the given drive function for each speaker, convolution of the impulse response vector calculated for each speaker with respect to one input acoustic signal, and outputting a weighted acoustic signal; For each speaker, the output time of the weighted sound signal is delayed by the time necessary to advance the distance between the speaker and the plurality of virtual sound sources at the speed of sound.
- a sound image reproduction program according to claim 6 causes a computer to function as the sound image reproduction device according to any one of claims 1 to 3.
- a sound image reproduction device a sound image reproduction method, and a sound image reproduction program that can be applied to a monaural sound source and that can impart directivity to a virtual sound source in a space.
- the present invention generates a virtual sound source in a circular shape in a space by using a linear speaker array using an input acoustic signal, and uses a circular harmonic expansion method that expands the acoustic signal into a circular harmonic series. It is characterized by giving directivity to the sound source.
- the present invention forms a circular array of virtual sound sources by generating a plurality of virtual sound sources in a circle on the front surface of the linear speaker array using the technique of Non-Patent Document 1, and Non-Patent Document 2 Using this technique, a virtual sound source having directivity is realized by giving different weights to the virtual sound sources in the circular array.
- FIG. 1 is a diagram illustrating a functional block configuration of an acoustic signal processing device 1 according to the first embodiment.
- the acoustic signal processing device (sound image reproduction device) 1 is a general computer including a processing device (not shown), a memory 10 and the like.
- a general computer implements the functions shown in FIG. 1 by executing an acoustic signal processing program (sound image reproduction program).
- the acoustic signal processing apparatus 1 receives an input acoustic signal I from a monaural sound source, uses a linear speaker array in which a plurality of speakers are arranged in a straight line, jumps to the front of the speakers, and has a virtual directivity. Realize the sound source.
- the acoustic signal processing device 1 converts the input acoustic signal I from the monaural sound source into an output acoustic signal O to each speaker of the linear speaker array in order to realize such a virtual sound source.
- the acoustic signal processing apparatus 1 includes a memory 10, a focal position determination unit 12, a filter coefficient determination unit 13, a convolution operation unit 14, an input / output interface (not shown), and the like. Composed.
- the input / output interface is an interface for inputting the input sound signal I from the monaural sound source to the sound signal processing apparatus 1 and outputting the output sound signal O to each speaker.
- the input / output interface inputs information about the coordinates of the virtual sound source and the direction of directivity realized by the acoustic signal processing device 1 to the acoustic signal processing device 1.
- the memory 10 stores focal coordinate data 11.
- the focal coordinate data 11 includes coordinate information for realizing a virtual sound source (hereinafter also referred to as a focal sound source) in the space.
- the focal point coordinate data 11 includes coordinates in an absolute coordinate system in which the line of linear speakers is the X axis and the front direction of the linear speaker is the Y axis.
- the center of a plurality of focal sound sources generated in a circle in the absolute coordinate system is defined as an origin O ′, and each axis parallel to the X and Y axes of the absolute coordinate system passing through the origin O ′ is represented by X ′.
- X ′ includes the coordinates of the relative coordinate system as the axis and Y 'axis.
- the focal position determination unit 12 receives information on the coordinates of the virtual sound source, the direction of directivity, and the target frequency, and outputs coordinates related to a predetermined number of necessary focal points.
- the focal position determination unit 12 determines the coordinate position of each focal sound source for generating a plurality of focal sound sources in a circular shape.
- the focal position determination unit 12 acquires the coordinate positions of a plurality of focal sound sources that are generated in a circle in the space of the absolute coordinate system, and uses the focal coordinate data 11 stored in the memory 10 to determine the relative coordinate system. Each polar coordinate is determined.
- the focal position determination unit 12 starts from the origin O ′ of the relative coordinate system.
- the relative coordinate system corresponding to the coordinate X s (x s , y s ) in the absolute coordinate system, where r s is the distance to the coordinate X s, and ⁇ s is the counterclockwise angle from the X ′ axis of the relative coordinate system.
- Determine the polar coordinate X s (r s , ⁇ s ).
- FIG. 2 is a diagram showing a focus position determination processing flow.
- FIG. 3 is a diagram illustrating an example of the coordinate position of the focal sound source in the absolute coordinate system and the relative coordinate system.
- step S11 the focal position determination unit 12 acquires each piece of information on the coordinates of the virtual sound source and the directionality of the virtual sound source generated in a circular shape in the space of the absolute coordinate system. Read data 11.
- the focal position determination unit 12 performs step S13 for each of the plurality of focal sound sources, and after step S13 has been performed for a predetermined number of focal sound sources, the process ends.
- the filter coefficient determining unit 13 processes the polar coordinates.
- the filter coefficient determination unit 13 receives the polar coordinates of all the focal sound sources output from the focal position determination unit 12, and also receives the coordinates of all the focal sound sources in the absolute coordinate system, and filters them in the frequency domain for each speaker. After designing, an impulse response vector given to each speaker is output by inverse Fourier transform.
- the filter coefficient determination unit 13 sets a drive function for each speaker in which a weight different from the others is given to a part of each focus sound source to a drive function for driving the speaker used to generate a focus sound source at the position of each focus sound source.
- the impulse response vector for each speaker is calculated by inverse Fourier transform.
- the filter coefficient determination unit 13 calculates an impulse response vector to be convoluted with the input acoustic signal I from each of the focal coordinates determined by the focal position determination unit 12 for each speaker of the linear speaker array.
- the filter coefficient determination unit 13 calculates a target frequency by an external input or the like, and applies the formula (3) and the formula (4) obtained by applying the formula (2) to the formula (1) to the speaker.
- the drive function to be given is calculated.
- X i (x i , y i ) is the coordinate position of the i-th speaker in the absolute coordinate system.
- X s (x s , y s ) is the coordinate position of the sth focal sound source in the absolute coordinate system.
- k ⁇ / c is the wave number.
- ⁇ is an angular frequency (2 ⁇ f).
- f is the frequency.
- c is the speed of sound.
- j is ⁇ ( ⁇ 1).
- H 1 (1) is a first-order first-class Hankel coefficient.
- g0 is ⁇ (2 ⁇
- W (r f , ⁇ f ) is a weight given to the focal sound source at the position (r f , ⁇ f ).
- S (2) (n, ⁇ ) is an n-th order circular harmonic series.
- J n (kr f ) is an nth-order Bessel function.
- the filter coefficient determination unit 13 calculates and uses the drive function of Expression (3) from Expression (1) and Expression (2).
- X i (x i , y i ) is the coordinate position of the i-th speaker in the absolute coordinate system.
- X s (x s , y s ) is the coordinate position of the s-th focal sound source in the absolute coordinate system (however, excluding X s of ⁇ Xs W (X s )).
- W (X s ) is a weight given to the focal sound source at the position X s .
- X s of W (X s) is the polar coordinate position of the s-th focus sound source in relative coordinates.
- the weight W (X s ) is given by Equation (4).
- X s (r s , ⁇ s ) is a polar coordinate position of the sth focal sound source in the relative coordinate system.
- S (2) (n, ⁇ ) is an n-th order circular harmonic series.
- J n (kr ′ f ) is an nth order Bessel function.
- X s used in the weight calculation of Expression (4) is a relative coordinate (r s , ⁇ s ) of each focal point with respect to the center of the circular array.
- the filter coefficient determination unit 13 circularly expands the n-order circular harmonic series obtained by circularly expanding the directivity characteristics of the focal sound source in advance with respect to a plurality of focal sound sources.
- the mode strength for each order is calculated by dividing the order by the two-dimensional green function, and the weighting coefficient of each focal sound source is obtained from the sum of the mode strengths for each order.
- a drive function consisting of Expressions (3) and (4) is derived.
- the two-dimensional Green function is known and can be uniquely defined.
- the filter coefficient determination unit 13 gives the i-th speaker among the speakers of the linear speaker array.
- a drive signal can be obtained.
- a virtual sound source having directivity can be realized by giving different weights to a plurality of focal sound sources based on information on the direction of directivity inputted externally.
- the filter coefficient determination unit 13 calculates this with respect to each speaker of the linear speaker array, and thereby obtains a drive signal with directivity to be given to each speaker.
- the filter coefficient determination unit 13 obtains an impulse response vector to be given to each speaker by performing an inverse Fourier transform on the drive function consisting of Expressions (3) and (4).
- FIG. 4 is a diagram showing a filter coefficient determination processing flow.
- step S21 the filter coefficient determination unit 13 acquires each focal point coordinate determined in the focal position determination process.
- the filter coefficient determination unit 13 repeats the processing from step S22 to step S26 to perform processing for calculating an impulse response vector for each speaker.
- the filter coefficient determination unit 13 initializes the impulse response vector of the target speaker to be processed with zero.
- the filter coefficient determination unit 13 initializes the impulse response vector in step S22, and then repeats the processing from step S23 to step S25 for each focal point.
- step S ⁇ b> 23 the filter coefficient determination unit 13 calculates a drive function composed of Expression (3) and Expression (4) for all desired target frequencies using the target focal coordinates to be processed.
- step S24 the filter coefficient determination unit 13 performs inverse Fourier transform on the drive function calculated in step S23 to obtain a time domain drive function.
- step S25 the filter coefficient determination unit 13 adds the time domain drive function acquired in step S24 to the impulse response vector.
- step S26 the filter coefficient determination unit 13 determines the impulse response vector at this time point as the impulse response vector to be given to the target speaker.
- step S23 to step S26 When the processing from step S23 to step S26 is completed for each speaker, the filter coefficient determination unit 13 ends the processing.
- or step S26 should just be performed with respect to each speaker, and may be performed in what order.
- the processing in steps S23 to S25 only needs to be performed for each focal point, and may be performed in any order.
- the convolution operation unit 14 convolves the impulse response vector with the input acoustic signal I, thereby giving output sound to each speaker.
- the signal O is calculated.
- the convolution operation unit 14 convolves an impulse response vector corresponding to each speaker with respect to one input acoustic signal I inputted to each speaker of the linear speaker array, and each of the weighted output acoustic signals O to the speakers. Output.
- the convolution operation unit 14 obtains a weighted output acoustic signal O for a predetermined speaker by convolving an impulse response vector corresponding to the speaker with the input acoustic signal I.
- the convolution operation unit 14 repeats the same processing for each speaker to obtain a weighted output acoustic signal O for each speaker.
- FIG. 5 is a diagram showing a convolution calculation process flow.
- the convolution operation unit 14 repeats the processing of step S31 and step S32 for each speaker of the linear speaker array.
- step S ⁇ b> 31 the convolution calculation unit 14 acquires the impulse response vector of the target speaker to be processed from the filter coefficient determination unit 13.
- step S ⁇ b> 32 the convolution operation unit 14 convolves the input acoustic signal I with the impulse response vector acquired in step S ⁇ b> 31 to acquire the output acoustic signal O.
- step S31 to step S32 When the processing from step S31 to step S32 is completed for each speaker, the convolution operation unit 14 ends the processing. Note that the processing in steps S31 to S32 only needs to be performed for each speaker, and may be performed in any order.
- the acoustic signal processing device (sound image reproduction device) 1 is a drive function used for generating a plurality of virtual sound sources in a circular shape, and a part of the virtual sound sources. Since a driving function to which a different weight is applied is used, a sound image reproduction device, a sound image reproduction method, and a sound image reproduction program capable of imparting directivity to a virtual sound source in a space can be provided.
- the acoustic signal processing device 1 convolves the impulse response vector for each speaker with each input acoustic signal, so that it can cope with a monaural sound source.
- FIG. 6 is a diagram illustrating a functional block configuration of the acoustic signal processing device 1 according to the second embodiment.
- the acoustic signal processing device (sound image reproduction device) 1 uses a filter calculation unit 15, a delay adjustment unit 16, and a gain multiplication unit 17 in place of the convolution calculation unit 14 shown in FIG. Is realized.
- the acoustic signal processing device 1 includes a memory 10, a focal position determination unit 12, a filter calculation unit 15, a delay adjustment unit 16, and a gain multiplication unit 17.
- the memory 10 and the focus position determination unit 12 are the same as those in the first embodiment.
- the filter calculation unit 15 uses the same method as in the first embodiment to calculate impulse response vectors calculated in advance using the equations (3) and (4) for each input acoustic signal I. Outputs convolutional and weighted acoustic signals. Similarly to the first embodiment, the filter calculation unit 15 calculates an impulse response vector in advance using Expression (3) and Expression (4) by the filter coefficient determination method illustrated in FIG.
- FIG. 7 is a diagram showing a filter calculation processing flow.
- step S41 the filter calculation unit 15 convolves the impulse response vector calculated in advance using Equation (3) and Equation (4) with the input acoustic signal I, and outputs a weighted acoustic signal.
- the delay adjustment unit 16 delays the output time of the weighted acoustic signal for each speaker of the linear speaker array by a time required to travel the distance between the speaker and the plurality of focal sound sources at the speed of sound. A delayed acoustic signal is output for each of the focal sound sources. The delay adjustment unit 16 calculates a delayed acoustic signal for all the focal points output by the focal position determination unit 12 using Equation (5). n is time.
- the gain multiplication unit 17 multiplies the delayed acoustic signals of the plurality of focal sound sources by the gains determined from the distances between the speakers and the plurality of focal sound sources, and outputs them to the speakers.
- An acoustic signal O is output.
- the gain multiplication unit 17 uses a delay adjustment unit 16 to obtain a gain obtained by dividing the distance between the focal point coordinates and the speaker array by the 3/2 power of the distance between the focal sound source and the speaker position for a predetermined speaker. By multiplying the obtained delayed acoustic signal, the output acoustic signal O is calculated.
- the “distance between the focal coordinate and the speaker array” is a difference between the value on the Y axis of the speaker array and the value on the Y axis of the focal coordinate when the speaker array is arranged on the X axis. .
- An output acoustic signal O for a predetermined speaker is obtained by Expression (6).
- the gain multiplication unit 17 calculates the output acoustic signal O for each speaker according to Expression (6).
- the delay adjusting unit 16 and the gain multiplying unit 17 perform processing of the delay adjusting unit 16 and the gain multiplying unit 17 that set a delay and a gain corresponding to the position of the speaker for a predetermined speaker of the linear speaker array, and output acoustic signals. Is generated.
- the delay adjustment unit 16 and the gain multiplication unit 17 obtain an output acoustic signal O for each speaker of the linear speaker array by performing the same processing while changing the speakers focused on this one after another.
- FIG. 8 is a diagram showing a flow of delay adjustment and gain multiplication processing.
- the acoustic signal processing device 1 performs the processing of step S51 and step S52 for each speaker of the linear speaker array.
- the delay adjustment unit 16 performs the process of step S51 for each focus.
- step S51 the delay adjusting unit 16 outputs a delayed acoustic signal obtained by delaying between the target speaker and the target focal point by the time of traveling at the speed of sound.
- step S52 the gain multiplication unit 17 multiplies the delayed acoustic signal for each focal point calculated in step S51 by the gain of the target speaker, and outputs the acoustic signal for the target speaker. Output O.
- step S51 and step S52 When the processing of step S51 and step S52 is completed for each speaker, the acoustic signal processing device 1 ends the processing.
- step S51 may be performed for each focus, and may be performed in any order.
- step S52 should just be performed with respect to each speaker, and may be performed in what order. Further, predetermined processing may be performed in parallel according to the processing environment or the like.
- the acoustic signal processing device (sound image reproduction device) 1 is a drive function used for generating a plurality of virtual sound sources in a circular shape, and among the virtual sound sources, Since a driving function having a weight different from others is used in part, it is possible to provide a sound image reproduction device, a sound image reproduction method, and a sound image reproduction program capable of imparting directivity to a virtual sound source in a space.
- the acoustic signal processing apparatus 1 since the acoustic signal processing apparatus 1 convolves the impulse response vector for each speaker with the inputted one acoustic signal, it can cope with a monaural sound source.
Abstract
Description
図1は、第1の実施形態に係る音響信号処理装置1の機能ブロック構成を示す図である。音響信号処理装置(音像再現装置)1は、処理装置(図示せず)、メモリ10などを備える一般的なコンピュータである。一般的なコンピュータが、音響信号処理プログラム(音像再現プログラム)を実行することにより、図1に示す機能を実現する。
第2の実施形態では、時間領域での波面合成を用いて、低演算量で仮想音源を多重極音源にする方法について説明する。
上記のように、本発明の第1及び第2の実施形態によって記載したが、この開示の一部をなす論述及び図面はこの発明を限定するものであると理解すべきではない。この開示から当業者には様々な代替実施の形態、実施例及び運用技術が明らかとなる。
10…メモリ
11…焦点座標データ
12…焦点位置決定部
13…フィルタ係数決定部
14…畳み込み演算部
15…フィルタ演算部
16…遅延調整部
17…ゲイン乗算部
Claims (6)
- 直線状に並べられた複数のスピーカを用いて仮想音源を空間内に生成する音像再現装置において、
複数の仮想音源を円状に生成するための各仮想音源の位置を決定する焦点位置決定部と、
前記各仮想音源の位置に仮想音源を生成するために用いるスピーカ駆動用の駆動関数に、前記各仮想音源のうち一部に他と異なる重みを付与したスピーカ毎の駆動関数を、逆フーリエ変換することにより、スピーカ毎のインパルス応答ベクトルを算出するフィルタ係数決定部と、
入力された1つの音響信号に対して前記スピーカ毎のインパルス応答ベクトルをそれぞれ畳み込み、それぞれの音響信号を前記複数のスピーカへそれぞれ出力する畳み込み演算部と、
を備えることを特徴とする音像再現装置。 - 直線状に並べられた複数のスピーカを用いて仮想音源を空間内に生成する音像再現装置において、
複数の仮想音源を円状に生成するための各仮想音源の位置を決定する焦点位置決定部と、
前記各仮想音源の位置に仮想音源を生成するために用いるスピーカ駆動用の駆動関数に、前記各仮想音源のうち一部に他と異なる重みを付与したスピーカ毎の駆動関数を、逆フーリエ変換することにより、予め算出されたスピーカ毎のインパルス応答ベクトルを、入力された1つの音響信号に対してそれぞれ畳み込み、重み付き音響信号を出力するフィルタ演算部と、
スピーカ毎に、前記スピーカと前記複数の仮想音源との間のそれぞれの距離を音速で進むのに必要な時間だけ前記重み付き音響信号の出力時間を遅延させ、前記複数の仮想音源のそれぞれについて、遅延音響信号を出力する遅延調整部と、
スピーカ毎に、前記スピーカと前記複数の仮想音源との間のそれぞれの距離から定まるゲインを、前記複数の仮想音源のそれぞれの前記遅延音響信号に乗じて出力するゲイン乗算部と、
を備えることを特徴とする音像再現装置。 - 前記スピーカ毎の駆動関数は、
前記複数の仮想音源に対して予め仮想音源の指向特性を円調和展開して得られたn次の円調和級数を、仮想音源に対して円調和展開された2次元グリーン関数で次数毎に除し、除した値を総和して各仮想音源の重み係数を求め、前記各仮想音源の重み係数と前記スピーカ駆動用の駆動関数とを加重平均した関数であることを特徴とする請求項1又は2に記載の音像再現装置。 - 直線状に並べられた複数のスピーカを用いて仮想音源を空間内に生成する音像再現方法において、
音像再現装置が、
複数の仮想音源を円状に生成するための各仮想音源の位置を決定するステップと、
前記各仮想音源の位置に仮想音源を生成するために用いるスピーカ駆動用の駆動関数に、前記各仮想音源のうち一部に他と異なる重みを付与したスピーカ毎の駆動関数を、逆フーリエ変換することにより、スピーカ毎のインパルス応答ベクトルを算出するステップと、
入力された1つの音響信号に対して前記スピーカ毎のインパルス応答ベクトルをそれぞれ畳み込み、それぞれの音響信号を前記複数のスピーカへそれぞれ出力するステップと、
を行うことを特徴とする音像再現方法。 - 直線状に並べられた複数のスピーカを用いて仮想音源を空間内に生成する音像再現方法において、
音像再現装置が、
複数の仮想音源を円状に生成するための各仮想音源の位置を決定するステップと、
前記各仮想音源の位置に仮想音源を生成するために用いるスピーカ駆動用の駆動関数に、前記各仮想音源のうち一部に他と異なる重みを付与したスピーカ毎の駆動関数を、逆フーリエ変換することにより、予め算出されたスピーカ毎のインパルス応答ベクトルを、入力された1つの音響信号に対してそれぞれ畳み込み、重み付き音響信号を出力するステップと、
スピーカ毎に、前記スピーカと前記複数の仮想音源との間のそれぞれの距離を音速で進むのに必要な時間だけ前記重み付き音響信号の出力時間を遅延させ、前記複数の仮想音源のそれぞれについて、遅延音響信号を出力するステップと、
スピーカ毎に、前記スピーカと前記複数の仮想音源との間のそれぞれの距離から定まるゲインを、前記複数の仮想音源のそれぞれの前記遅延音響信号に乗じて出力するステップと、
を行うことを特徴とする音像再現方法。 - 請求項1乃至3のいずれかに記載の音像再現装置としてコンピュータを機能させることを特徴とする音像再現プログラム。
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