JP2005223646A - Sound adjustment console - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a sound adjustment console carrying out output control of a direct sound and a reflected sound in matching with an actual sound pressure without the need for configuring a large scale system and the need for expense in many editing times and much labor. <P>SOLUTION: The sound adjustment console 1 interlocks with each other: control of a sense of distance from a sound source to a listener; control of localizing a direction of the sound source depending on the sound listened to by the listener; and control of generating a reflected sound resulting from the reflection of the direct sound emitted from the sound source, so as to control outputs of a plurality of speakers arranged around the listener. The console 1 is provided with: a distance filter 9; a direct sound side variable attenuator 11; a reflection sound side variable attenuator 13; a direction localizing unit 15; a reverberation circuit 17; an interlocking control unit 23; and an operating unit 25. <P>COPYRIGHT: (C)2005,JPO&NCIPI

Description

  The present invention relates to a sound adjustment console that performs various digital signal processing on an input acoustic signal.

  Conventionally, a sound adjustment console (see, for example, Patent Document 1) is generated from a sound source and a sense of distance from the sound source to a listener by performing digital signal processing on a direct sound (acoustic signal) emitted from the sound source. The direction of the direct sound, the reflected sound reflected by the reflection surface forming the acoustic space, and the direct sound and ratio are controlled.

  The sense of distance from the sound source to the listener is controlled by a distance attenuation control element (such as a filter) and a reflected sound control element (such as a reverb circuit), and the direction of the sound source is controlled by a direction localization control element. Yes. That is, the sense of distance from the sound source to the listener and the direction of the sound source are controlled individually (separately).

  In the conventional audio control console, the distance control and the direction control are performed by adding time information called a time code to the control result (recording on the time code) and controlling each control result. By repeating the above, desired characteristics were obtained.

  In other words, when expressing the movement of a sound source using a conventional audio control console, a filter function using a distance attenuation control element, a level attenuation function using a variable attenuation control element (attenuator, etc.), a direction localization function using a direction localization control element, a reflected sound Since it was necessary to control the reverb control function (adjustment of the reverb control amount) by the control element at the same time, or simultaneously control multiple channels (input channels that control audio signals emitted from other sound sources), the automation function ( It was realized using time code automation.

  Furthermore, when expressing and storing the movement of a sound source using a conventional audio adjustment console, the adjustment amount by the filter function, the level attenuation amount by the level attenuation function, the control result by the direction localization function, and the reverb control amount by the reverb control function are simultaneously displayed Alternatively, it is necessary to store a plurality of channels simultaneously as automation data of the same time code.

  In addition, in the conventional audio adjustment console, as a grouping, there is a function in which a fader of another channel is linked to an arbitrary fader. It is possible to move the sound source at the same time. However, although this function can maintain any time linkage, it is impossible to link each fader in the time axis direction, and time code automation data is required to realize this linkage.

By the way, a device for binaural sound generation using HRTF (Head Related Transfer Function) reproduces a three-dimensional sound field by listening to headphones, and in order to generate HRTF, the position of a listener is determined. An operation panel for setting a three-dimensional sound field is displayed at the center.
JP 2001-134281 A (paragraphs 0015 to 0034, FIG. 1)

  However, the automation function in the conventional audio control console is complicated to operate because the sense of distance from the sound source to the listener and the direction of the sound source are controlled individually (separately). This requires a large system configuration for configuring the audio adjustment console. These functions, ie, filter function, level attenuation function, direction localization function, and reverb control function, are provided in a single voice adjustment console, and each function can reproduce direct sound and reflected sound at an arbitrary timing. It was impossible.

  Also, when expressing and storing the movement of a sound source using a conventional audio adjustment console, parameters (volume level, output timing of output sound from each speaker, reflected sound) obtained by controlling one function 1) must be stored as automation data one by one, and there is a problem that much editing time and labor (labor) are required. Furthermore, each parameter must be synchronized with the time code, and there is a problem that the operation becomes complicated. In addition, in the conventional audio adjustment console, each function is difficult to understand for the operator who operates the audio adjustment console. In other words, when editing sound using a conventional audio adjustment console, there is no sense of unity in the relationship between the distance between the control amount of the filter and the like handled by the operator of the audio adjustment console, that is, the operator The relationship between the sensory changes in the sound to be operated and the display on the display screen was sparse.

  In effect devices that generate recent stereophonic sound (3D sound), parameters obtained as a result of controlling the filter function, level attenuation function, direction localization function, and reverb control function are displayed on the xy plane. There are things that can be created with the graph creation tool, but as each parameter is created, it becomes unclear how these parameters are controlled, so if you want to fine-tune the parameters instantly Is unsuitable.

  Furthermore, conventional binaural sound generation devices have a visually easy-to-understand operation panel, which is excellent for expressing the direction of the sound source, but does not limit the position of the output speaker. There is a problem that it is impossible to visually recognize how much sound is output from the speaker.

  Therefore, the present invention solves the above-described problems, eliminates the need for a large-scale system configuration, and controls output of direct sound and reflected sound that match actual sound pressure without spending much editing time and labor. An object of the present invention is to provide a sound adjustment console that can be performed.

  In order to solve the above-mentioned problem, the voice adjustment console according to claim 1 is a voice adjustment that performs control of a sense of distance from a sound source to a listener and control for localizing a direction of the sound source from a sound heard by the listener. The table is configured to include an operation unit, an analysis unit, and a storage unit.

  According to such a configuration, the audio adjustment console inputs position information indicating the position of the sound source by the operation means. The position information indicating the position of the sound source is given by the x and y coordinates on the two-dimensional plane. Subsequently, the voice adjustment console analyzes the control amount of the non-control unit in which the sense of distance and the direction localization are controlled by the analysis unit. This non-control means is a distance sense control means (distance filter) for controlling the sense of distance, a direct sound control means (direct sound side variable attenuator) for controlling the direct sound emitted from the sound source, and a reflected sound reflected by the direct sound. Is reflected sound control means (reflected sound side variable attenuator). The non-control means is not limited to these, and may be various means (circuits) on the sound adjustment table, for example, a head amplifier, a pan circuit, or the like. Further, the analysis means converts the position information indicating the position of the sound source into polar coordinates, and the control amount (for example, parameter) corresponding to the magnitude component is, for example, analyzed data in which the relationship between each parameter is set in advance. It is determined by referring to the table. And the audio | voice adjustment console memorize | stores the controlled variable analyzed by the analysis means by the memory | storage means.

  When the sound source is moved, the position information is changed (rewritten) by the operation means, or the locus of the sound source is drawn using an interface such as a touch panel. For example, the drawn trajectory is divided into arbitrary intervals (a fixed interval such as a time code or whenever it changes), and the control amount (parameter) of the non-control means at each coordinate (point) is obtained, and a number is assigned to each And stored in the storage means. It is possible to control the sound from the sound source by outputting the control amount analyzed at this time to the non-control means. Further, the control by the non-control means can be reproduced by tracing the drawn trace using an interface such as a touch panel as an operation means.

  The voice adjustment console according to claim 2 is a voice adjustment console that adjusts at least one of volume and sound quality for each of a plurality of input audio signals, and includes a non-control means, a control data storage means, and an interlocking operation. Means and interlock control means.

  According to such a configuration, the audio adjustment console designates the control data stored in the control data storage means by the interlock operation means. At this time, the voice adjustment console reads the control data designated by the interlock operation means by the interlock control means at an arbitrary speed from the control data storage means, and adjusts at least one of the volume and the sound quality. Control means in conjunction. The interlocking operation means may be configured with an actual fader or the like, or may be configured with a virtual fader or the like on the display screen.

  For example, control data of arbitrary non-control means (distance control means, direct sound control means, reflected sound control means, direction localization control means) is designated by the interlock operation means. Since the non-control means is read out by the interlock control means at an arbitrary speed, the state where the sound source moves at an arbitrary speed can be reproduced by one interlock operation means and the interlock control means. In addition, by holding time data in this interlocking operation means, it is also possible to reproduce the sound from the sound source at a constant speed triggered by a trigger such as a switch. Further, when the time data is a time code, time code automation is possible as in the conventional case.

  The non-control means may not be limited to the sense of distance control means, the direct sound control means, the reflected sound control means, or the direction localization control means, and the non-control means may share a plurality of channels. In this case, a setting control amount is added to the conventional grouping function, and sounds from a plurality of sound sources can be reproduced at an arbitrary speed by one adjusting means.

  The voice adjustment console according to claim 3 is the voice adjustment console according to claim 2, wherein the positional relationship between the position of the listener and the position of the speaker that outputs the sound controlled by the non-control means, the interlock control. It is characterized by comprising display means for displaying the control result by means and the locus of movement of the sound source.

  According to such a configuration, the voice adjustment console displays the positional relationship between the position of the listener and the position of the speaker that outputs the sound controlled by the non-control means, for example, the position of the listener (listening position). A layout diagram in which eight loudspeakers are arranged at equal intervals on a circle having a center point (reference point) is displayed. Further, the display means displays the control result by the interlock control means as a line graph, and further, the locus of the sound source that changes based on the position information input by the operation means, or the locus of the sound source drawn by the touch panel or the like. Display.

  The sound adjustment console according to claim 4 performs a control of a sense of distance from the sound source to the listener and a control to localize the direction of the sound source from the sound heard by the listener, and a direct sound emitted from the sound source. Is a voice adjustment console that mixes the reflected sound reflected by the direct sound and the direct sound, the operation means, the distance attenuation control means, the direction localization control means, the reflected sound control means, the interlock control means, and the output means, It was set as the structure provided with.

  According to such a configuration, the audio adjustment console inputs position information indicating the position of the sound source by the operation means. Subsequently, the voice adjustment console reflects the high frequency attenuation by the distance filter, the direct sound emitted from the sound source, and the direct sound by the distance attenuation control means based on the position information input by the operation means. Controls the sense of distance caused by adjusting the direct ratio with the reflected sound. That is, this sense of distance is controlled based on the high-frequency attenuation and the direct ratio that is the ratio of the direct sound and the reflected sound (indirect sound).

  And this audio | voice adjustment console performs control which localizes the direction of a sound source based on position information by a direction localization control means. This localization of the direction of the sound source is achieved by changing the output locations of the direct sound and the reflected sound output from a plurality of speakers, and adjusting the volume levels of the direct sound and the reflected sound. In addition, the voice adjustment console generates a reflected sound from the direct sound by the reflected sound control means. This reflected sound changes depending on the reflection coefficient of the reflecting surface of the acoustic space where the sound source is arranged.

  Further, the voice adjustment console uses the interlock control means to link the control result by the distance attenuation control means, the control result by the direction localization control means, and the control result by the reflected sound control means, so that the direct sound and the reflected sound are controlled. Controls output timing, output location, and volume level. That is, by this interlock control means, it is determined at what timing and how much the direct sound and the reflected sound are mixed and output from which speaker and at what volume level. Thereafter, the voice adjustment console outputs the direct sound and the reflected sound controlled by the interlock control means to the plurality of speakers by the output means.

  According to the first aspect of the present invention, it is not necessary to make a large-scale system configuration by using the control amount analyzed by the analysis means for the distance and direction adjustment work that has been conventionally performed sensibly. It is possible to control the output of the direct sound and the reflected sound that match the actual sound pressure without consuming editing time and labor.

  According to the invention described in claim 2, since the change of the control data designated by the interlock operation means is interlocked at an arbitrary speed by the interlock control means, the interlocking work realized by the conventional time code automation is performed. Can be easily realized.

  According to the third aspect of the present invention, the display means displays the positional relationship between the position of the listener (listening position) and the position of the speaker, the control result by the interlock control means, and the locus of the sound source. The operator who operates can visually grasp the listening position, the speaker, the control result, and the sound source, and is easy to understand sensuously, so that the usability is improved.

  According to the fourth aspect of the present invention, the parameters for controlling the distance attenuation control means, the direction localization control means and the reflected sound control means calculated from the locus drawn by the operation means are linked by the linkage control means. Therefore, the efficiency of editing work is improved.

Next, embodiments of the present invention will be described in detail with reference to the drawings as appropriate.
FIG. 1 is a block diagram of an audio adjustment console. As shown in FIG. 1, the sound adjustment console 1 performs control of a sense of distance from a sound source to a listener, control of the direction of the sound source, and control of the direct ratio, in combination with a head amplifier. 3, an equalizer 5, a fader 7, a distance filter 9 (distance attenuation control means), a direct sound side variable attenuator 11, a reflected sound side variable attenuator 13, and a direction localizer 15 (direction localization control means). A reverb circuit 17 (reflected sound control means), a mixing means 19, a first storage unit 21, an interlock control unit 23 (interlock control means), an operator A25 (25a) (operator means), and an operator. B25 (25b) (interlocking operation means), display A27 (27a) (display means), display B27 (27b) (display means), analyzer 29 (analysis means), analysis database 31, Second storage unit 33 (storage means) and third storage unit 3 And a (control data storage means) and.

  The head amplifier 3 is composed of a plurality of transistors and the like, and amplifies an inputted acoustic signal (original sound and direct sound) to a specified level (increases the energy of the signal).

  The equalizer 5 adjusts (corrects or changes) the timbre (frequency characteristic) of the acoustic signal (direct sound) amplified by the head amplifier 3.

  The fader 7 is composed of a variable resistor or the like, and adjusts the volume level of the acoustic signal whose tone color has been adjusted by the equalizer 5 by changing the resistance value of the variable resistor (overall level control). . That is, attenuation (sound distance attenuation) over the entire frequency band (all audio frequency components) of the acoustic signal is performed.

  The distance filter 9 (distance attenuation control means) attenuates the volume level of the high frequency component in the band of the acoustic signal with respect to the acoustic signal attenuated by the fader 7 so as to correspond to a desired sound image distance. . The acoustic signal attenuated by the distance filter 9 is branched into two. The volume level is adjusted based on a control signal (parameter [control amount]) output from the interlock control unit 23, which will be described in detail later.

Returning to FIG. 1, the description of the configuration of the audio adjustment console 1 will be continued.
The direct sound side variable attenuator 11 is based on the control signal (parameter [control amount]) output from the interlock control unit 23, and is one volume level (direct sound) of the acoustic signal attenuated and branched by the distance filter 9. (Side level) is attenuated. The direct sound side variable attenuator 11 attenuates an acoustic signal that is a direct sound as it is.

  The sense of distance by the direct sound side variable attenuator 11 can be defined by the sound pressure that is the sound pressure, as shown by the “distance” straight line shown in FIG. That is, in the “distance” in FIG. 3, when the position on the right side of “near” shown in the right end portion in FIG. 3 is the listener's position, the “infinity point” shown in the left end portion in FIG. When a sound source exists by “100 m”, the sound pressure is −30 dB constant. Further, when the sound source exists from “100 m” to “3 m”, the sound pressure increases, and when the sound source exists at the “3 m” point, the sound pressure becomes 0 dB. Then, when the sound pressure exists from “3 m” to “near”, the sound pressure increases, and when the sound source exists at the “near” point, the sound pressure becomes +10 dB.

  In addition, in “fader” (sound level), when the sound source is at an infinite point from the listener, the sound pressure becomes −∞ (infinity) dB, and when the sound source approaches 100 m from the listener. The sound pressure becomes 0 dB, and the sound pressure remains 0 dB even when the sound source approaches the vicinity of the listener.

  Furthermore, in “auxsend” (reflected sound feed), the sound source is constant from the point of infinity until it reaches 3 m of the listener, and as the sound source approaches from 3 m to the listener, the sound pressure decreases, In the vicinity, the sound pressure is −∞ (infinity) dB.

  For reference, the distance from the listener (listening position) to the sound source is determined by the volume level of each of the direct sound, reflected sound (initial reflection), and reverberant sound, and the time difference until the listener is reached. .

  The reflected sound side variable attenuator 13 is based on the control signal (parameter [control amount]) output from the interlock control unit 23, and the other volume level (reflected sound) of the acoustic signal attenuated and branched by the distance filter 9. (Side level) is attenuated. The reflected sound side variable attenuator 13 is provided in a feed (path) to the reverb circuit 17 for attenuating an acoustic signal that is a direct sound and generating a reflected sound.

  The direction localizer 15 (direction localizing means) is a channel for outputting the acoustic signal attenuated by the direct sound side variable attenuator 11 based on the control signal (parameter [control amount]) output from the interlock control unit 23. It distributes and controls the direction localization of the sound source.

  The reverb circuit 17 (reflection sound control means) is an acoustic signal attenuated by an arbitrary attenuation amount by the reflection sound side variable attenuator 13 based on the control signal (parameter [control amount]) output from the interlock control unit 23. A reflected sound (indirect sound) is generated (created) from the sound. This reflected sound (indirect sound) is a “sounding signal” when an acoustic signal is emitted in a specific space and reflected by a wall surface (reflecting surface) constituting the space. In the reverb circuit 17, the generated reflected sound is assigned to each output channel (in this case, each bus line connected to each speaker (not shown) when output).

  In this embodiment, the reverb circuit 17 is integrally incorporated in the sound adjustment console 1. However, for example, the reverb circuit 17 may be configured as an external device (effector). Further, as shown in FIG. 1, the reverb circuit 17 has one input and multiple outputs. Or it becomes multi-input multi-output. In the case where the reverb circuit 17 has multiple inputs and multiple outputs, a direction localizer 15 (a direction localizer on the reflected sound side) is interposed before the reverb circuit 17. In this case, the direction determiner on the reflected sound side can be interlocked with the direction determiner 15 on the direct sound side, and the control signal (parameter [control amount]) input from the interlock control unit 23 is constant (constant value). is there.

  Here, the realism of the reverberation (generating a reflected sound) by the reverb circuit 17 and the sound field actually reproduced will be described (see the Sawaguchi Masao edition surround production handbook).

  The reverb circuit 17 generates reflected sound according to the acoustic characteristics of various spaces. This acoustic characteristic is the total amount of energy reflected by the reflective surfaces such as the ceiling, floor, and wall that form the space (sound field), that is, the original sound (direct sound) emitted from the sound source hits the reflective surface and is reflected. In addition, it can be defined based on the total amount it reflects. That is, the acoustic characteristics vary depending on the shape and complexity of the space (sound field), the state of the reflection surface, the sound absorption characteristics of the reflection surface, the distance from the sound source to the listener, and the direction.

  In addition, since a specific reflection pattern exists in a specific space (sound field), it is possible to output a digital acoustic signal processed by the audio adjustment console 1 from, for example, a 5-channel speaker. By generating a reflected sound that approximates the reflection pattern as much as possible, it is possible to give a high sense of realism to the listener who listens to the digital sound signal.

  In further explanation, it is generally said that a human can sense a sense of distance (a sense of perspective from a sound source to a listener) from a reflected sound component of the sound. In a general indoor space, when a direct sound reaches the human ear, the human detects the direction of arrival of the direct sound by a slight time delay and a level difference generated between both ears. When the reflected sound component reaches the human ear with a delay of about 10 msec after the direct sound arrives, the human views the reflected sound component as distance information and obtains a sense of distance.

Returning to FIG. 1, the description of the configuration of the audio adjustment console 1 will be continued.
The mixing means 19 mixes the acoustic signal (direct sound) assigned to each output channel by the direction determiner 15 and the reflected sound (indirect sound) generated by the reverb circuit 17 and assigned to each output channel. Output.

  The 1st memory | storage part 21 is comprised by the non-volatile memory etc., and memorize | stores the input acoustic signal (original sound, direct sound). The first storage unit 21 sends the stored acoustic signal to the head amplifier 3 according to the control signal output from the interlock control unit 23. That is, by storing the acoustic signal in the first storage unit 21, it is not necessary to input (acquire) a certain pattern of acoustic signals one by one.

  The interlock control unit 23 (interlock control means) operates the operator B25 (25b) by the operator of the voice adjustment console 1, and based on the operation signal output from the operator B25 (25b), the distance filter 9 Control of the sense of distance, control of direct ratio by the direct sound side variable attenuator 11 and reflected sound side attenuator 13, control of generation of reflected sound by the reverb circuit 15, and direction localization of the sound source by the direction localizer 15. The control is performed in conjunction with the above control.

  That is, the interlock control unit 23 is displayed on the display unit B27 (27b) by using the lever-like device of the operator B25 (25b) or a device such as a keyboard or a mouse. The position information of the sound source (position information including time change (x, y coordinates)) is acquired by moving the slider, the display switch, etc., and the distance filter 9, the direct sound side variable attenuator 11, the reflected sound side The parameters to be output to the attenuator 13, the direction localizer 15, and the reverb circuit 17 are calculated, and the plurality of parameters are output as control signals, whereby each of them is linked and controlled. That is, the result of the interlock control performed by the interlock control unit 23 is displayed on the display device B27 (27b).

  Further, the interlock control unit 23 outputs a control signal to the first storage unit 21, causes the head amplifier 3 to output the acoustic signal stored in the first storage unit 21, and outputs the stored acoustic signal. It is intended to be reused.

  The operator A25 (25a) (operating means) is composed of a lever-like device such as a joystick and devices such as a keyboard and a mouse, and inputs two-dimensional coordinates indicating the position of the sound source and a locus of movement of the sound source. Is to do. When the operator A25 (25a) is operated by an operator who operates the sound adjustment console 1, an operation signal is output from the operator A25 (25a) to the analyzer 29.

  The operation element B25 (25b) (interlocking operation means) is configured by a lever-like device such as a fader, or a device such as a keyboard or a mouse, and a control amount (parameter) stored in the third storage unit 35. Is operated (changed, etc.).

  Display A27 (27a) (display means) is based on the operation signal output from operation element A25 (25a), and the position of the listener and the positions of a plurality of speakers arranged around the listener. Is displayed on the display screen, and when the sound source moves, the locus of the movement of the sound source is displayed. By displaying in this way, it is possible to adjust the sense of distance and the direction without worrying about the control amounts (parameters) of the filter, fader, etc. that were normally operated by the operator.

  The display device B27 (27b) (display means) displays each control amount (parameter) as a line graph or the like based on the operation signal output from the operation element B25 (25b). This line graph displays what kind of control data is stored in the third storage unit 35 as to the direction of the sound source, the level of the reflected sound, the level of the distance filter 9, and the overall level. Is. Further, these line graphs and the like can be rewritten by editing them using the operation element B25 (25b) such as a mouse.

  The display B27 (27b) displays a plurality of sliders (adjustment switches (knobs) to be slid) for adjusting the volume level, attenuation level, and distance of the direct sound and the reflected sound. This slider (time slider) is set in association with each of automation data for automatically controlling the sense of distance of direct sound and indirect sound, direction localization, generation of reflected sound, etc. according to time slider data (details) Will be described later with reference to FIG. 4). By adjusting (sliding) this slider (time slider) using the mouse or the like of the operation element B25 (25b), the interlock control unit 23 controls the sense of distance, the control of orientation, and the direct sound and reflection. The direct ratio with the sound can be adjusted with the passage of time.

  In addition, this display B27 (27b) improves the operability by displaying the positional relationship between a plurality of speakers and the direction in which the sound source is localized in order to display the direction of the sound source more clearly. It is.

  The analyzer 29 (analyzing means) includes two-dimensional coordinates (x, y) indicating the position of the sound source (sound image to be localized) input from the operator A25 (25a) and the analysis data stored in the analysis database 31. Based on the data (angle function data, reference point data), the sound source direction localization and magnitude (volume level magnitude) are calculated, and the calculated result (parameter [control amount]) is stored in the second storage unit. 33 or the third storage unit 35, and includes an angle function setting unit and a control amount calculation unit.

The angle function setting means sets an angle function based on the input two-dimensional coordinates (x, y), the angle function data stored in the analysis database 31 and the reference point data. This angle function uses an angle θ formed by a reference line (x-axis, y-axis) orthogonal to the reference point and a straight line connecting the reference point and the input two-dimensional coordinates as a variable.
In this angle function, r is the distance from the reference point to the two-dimensional coordinate.

  The angle function data is a sample of the angle function defined by the angle θ. The reference point data is represented by (x, y) coordinates indicating the position of the listener.

  The control amount calculating means is based on the angle function set by the angle function setting means, the speaker position data (speaker position information) stored in the analysis database 31, and the level calculation corresponding data. The amount of control to each non-control means (distance filter 9, direct sound side variable attenuator 11, reflected sound side variable attenuator 13 and direction localizer 15) output via 23 is calculated. In this control amount calculation means, in order to localize the phantom sound image (reproduction sound field), parameters (control amounts) are calculated so that output sound is output from one or two speakers.

  The level calculation correspondence data defines the relationship between the two-dimensional coordinates (x, y) indicating the position of the input sound source and the volume level.

  In this analyzer 29, when the position of the sound source is moved, the angle function setting means sequentially reads the angle function data from the analysis database 31 and resets the angle function as the sound source moves. Then, the speaker position data and the level calculation corresponding data are read out by the control amount calculation means, and each non-control means (distance filter 9, direct sound side variable attenuator 11, reflected sound side variable attenuator 13 and direction localization is read out. The control amount to the device 15) is reset and output to the second storage unit 33 or the third storage unit 35.

  The analysis database 31 includes a nonvolatile memory, a hard disk, and the like, and stores angle function data, reference point data, speaker position data, and level calculation correspondence data.

  The second storage unit 33 (storage means) is configured by a nonvolatile memory, a hard disk, or the like, and stores parameters (control amounts) calculated by the analyzer 29. The parameter (control amount) stored in the second storage unit 33 may be output to the interlock control unit 23 in accordance with the operation signal output from the operator A25 (25a).

  The third storage unit 35 (control data storage means) is configured by a nonvolatile memory, a hard disk, or the like, and stores control data that is a preset (externally input) control amount. The parameters (control data) stored in the third storage unit 35 are output to the interlock control unit 23 in accordance with the operation signal output from the operation element B25 (25b).

  According to the voice adjustment console 1, position information indicating the position of the sound source is input by the operator A25 (25a), and the position information input by the operator A25 (25a) is input to the control signal (parameter) by the analyzer 29. Based on this control signal (parameter [control amount]), the interlock control unit 23 causes the distance filter 9, the direct sound side variable attenuator 11, the reflected sound side variable attenuator 13 and the like. Is adjusted by adjusting the volume level over the entire frequency band and the volume level of the high frequency component in the band, and the distance produced by the direct sound emitted from the sound source and the reflected sound reflected by the direct sound The feeling is controlled. The direction localizer 15 controls the direction localization of the sound source based on the control signal (parameter [control amount]), and the reverb circuit 15 generates reflected sound from the direct sound.

  For this reason, since the parameter (control amount) analyzed by the analyzer 29 is controlled by the interlock control unit 23 in conjunction with each non-control means, there is no need for a large system configuration, and the operator A25 (25a) Since the operator operates and the control result is displayed on the display A27 (27a) in a manner that is easy to understand, the direct sound that matches the actual sound pressure can be obtained without consuming much editing time and labor. And the output control of the reflected sound can be performed.

  Further, according to the voice control console 1, when the sound source moves from an arbitrary point to another arbitrary point at an arbitrary speed, the slider displayed on the display 27B (27b) is moved to the operation element B25 (25b). Since the output sound (direct sound and indirect sound) is controlled by the interlock control unit 23 by operating with a mouse or the like, the direct sound that matches the actual sound pressure is saved without spending much editing time and labor. And the output control of the reflected sound can be performed.

  Furthermore, according to the voice adjustment console 1, the display A27 (27a) is a line graph that graphs the positional relationship between the position of the listener and the positions of the plurality of speakers and the parameters that are the analysis results by the analyzer 29. Since the sound source trajectory that changes based on the position information and the trajectory of the drawn sound source are displayed, it is easy for the operator of the voice adjustment console 1 to feel the sound source operation, and much editing time and labor are spent. Therefore, it is possible to control the output of the direct sound and the reflected sound that match the actual sound pressure.

(Voice adjustment console operation)
Next, the operation of the audio adjustment console 1 will be described with reference to the flowchart shown in FIG. 2 (see FIG. 1 as appropriate).
First, an acoustic signal is input to the head amplifier 3 of the audio adjustment console 1 (S1). Then, the head amplifier 3 amplifies the input acoustic signal to a specified level (S2). Subsequently, the audio adjustment console 1 attenuates the high frequency audio frequency component of the acoustic signal by the equalizer 5 (S3). Further, the audio adjustment console 1 attenuates all audio frequency components by the fader 7 (S4).

  Subsequently, two-dimensional coordinates indicating the position of the sound source are input to the voice adjustment console 1 by the operator A25 (25a) (S5). Then, the voice adjustment console 1 reads the reference point data from the analysis database 31 by the angle function setting means of the analyzer 29 (S6) and also reads the angle function data (S7). At this time, the display A27 (27a) displays a graphic indicating the relationship between the position of the listener and the position of the speaker and (x, y) coordinates indicating the position of the sound source. ing.

  Then, the voice adjustment console 1 sets the angle function by the angle function setting means of the analyzer 29 (S8), and the locus indicating the movement of the sound source is detected from the operator A25 (25a), that is, whether the sound source has moved. It is determined whether or not an operation signal indicating that it has been drawn has been output (S9). When it is determined that the sound source has moved (S9, Yes), the audio adjustment console 1 changes the display of the display A27 (27a), that is, displays the locus of the movement of the sound source (S10), S7. Return to (read out the angle function data again).

  When it is determined that the sound source has not moved (S9, No), the audio adjustment console 1 calculates the control amount by the control amount calculation means of the analyzer 29, and the volume level (all audio frequency components) by the distance filter 9 The direct sound component by the direct sound side variable attenuator 11 and the component that becomes the reflected sound by the reflected sound side variable attenuator 13 are linked and adjusted (linked control) (S11).

  Then, the sound adjusting console 1 mixes and outputs the direct sound and the reflected sound output from the direction localizer 15 and the reverb circuit 17 by the mixing unit 19 (S12).

(About automation data and control signal [parameter])
Next, referring to FIG. 4 and FIG. 5, the automation data and the control signal output by the interlock control unit 23 when the sound source moves (when the movement of the sound source is drawn by the operation element B25 (25b)). The relationship with the parameter [control amount]) will be described (see FIG. 1 as appropriate).

As shown in FIG. 4, the slider controlled by the operation element B25 (25b) and displayed on the display unit B27 (27b) includes a plurality (N in this case) of automation data. The data is assigned time slider data in which the same numbers are associated with each other.
That is, the automation data is changed by changing (sliding) the “time slider” displayed on the display device B27 (27b).

The automation data may be an arbitrary parameter (control amount). Further, this automation data may extend over a plurality of channels instead of a single channel. Further, the automation data may be associated with a time code, and in this case, it can be handled as normal time code automation data.
Further, the automation data may be reproduced at a constant speed by a trigger such as an arbitrary switch on the audio adjustment console 1.

  Further, as shown in FIG. 5, when the listener's listening position P and the locus of the sound source Q are displayed on the display A27 (27a), that is, using the mouse or the like of the operator A25 (25a), When the locus of the sound source Q is drawn, the output of the reflected sound side variable attenuator 13 becomes a function of the distance r, attenuates from a constant transition, and the output of the distance filter 9 also becomes a function of the distance r. From a certain distance, it becomes constant (constant), and the output of the fader 7 also becomes a function of the distance r and becomes constant (constant).

  In addition, the output to the direction determiner 15 is a function of the angle θ, and a speaker that outputs based on the angle θ is selected. When the angle θ is 0 °, 45 °, 90 °, 135 °, 180 °, 225 °, 270 °, 315 °, one speaker is selected from a plurality of speakers. When the angle θ is other than that, two neighboring (adjacent) speakers are selected from the plurality of speakers, and the output sound to be output is calculated as a partial pressure between the two speakers.

  For example, the automation data created when the position information input from the operator A25 (25a) is drawn as a trajectory as shown in FIG. 5 is used when an arbitrary interval occurs in the time code or the like. Is recalculated every time there is a change, and stored in the second storage unit 33.

(About the display image displayed on the display)
Next, a display image (slider or the like) displayed on the display A 27 (27a) or the display B 27 (27b) will be described with reference to FIGS. 6 to 9 (see FIG. 1 as appropriate).
6 to 9 are display images displayed on the display 23, and these display images have the same function as the “window” employed in the personal computer. FIG. FIG. 7 shows an auto pan window (window displayed on the display unit B27 (27b)), and FIGS. 8 and 9 show a trace window (display unit A27 (27a)). Window to be displayed).

  As shown in FIG. 6, in the main window, the name of the acoustic signal file being edited with the “FILE” name is displayed, “AUX” for adjusting the feed of the reflected sound, “DISTANCE” for adjusting the sense of distance, A “LEVEL” time slider that adjusts the attenuation level, an “AUTO” jog dial that automatically performs pan control, and “START” that indicates the start time are displayed.

  As shown in FIG. 7, in the auto pan window, the “AUTO” jog dial that automatically controls pan, “POSITION” indicating the position of the sound source, “AUX” indicating the transmission of the reflected sound, and from the sound source to the listener A line graph of “DISTANCE” indicating the distance and “LEVEL” indicating the attenuation level is displayed.

  As shown in FIG. 8, in the trace window, the listener, eight speakers, and the locus of the sound source are displayed around the listener's position. In FIG. 9, the listener's position, The track of the sound source is displayed.

  As mentioned above, although embodiment of this invention was described, this invention is not limited to the said embodiment. For example, although the present embodiment has been described as the audio adjustment console 1, the processing of each component of the audio adjustment console 1 can also be understood as a program (audio adjustment program) described in a general-purpose computer language. In this case, the same effect as that of the audio adjustment console 1 can be obtained.

It is a block diagram of the audio | voice adjustment console which concerns on embodiment. It is the flowchart explaining operation | movement of the audio | voice adjustment console shown in FIG. It is the figure explaining the relationship between the listening position from a sound source, and sound pressure. It is a figure explaining automation data and a control signal (parameter). It is a figure explaining the control signal (parameter) by the display of a display, and the analysis result of an analyzer. It is a figure explaining the window (main window) displayed on a display. It is a figure explaining the window (autopan window) displayed on a display. It is a figure explaining the window (trace window) displayed on a display. It is a figure explaining the window (trace window) displayed on a display.

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 Voice adjustment table 3 Head amplifier 5 Equalizer 7 Fader 9 Distance filter (distance attenuation control means)
11 direct sound side variable attenuator 13 reflected sound side variable attenuator 15 direction localizer (direction localization control means)
17 Reverb circuit (reflection sound control means)
19 Mixing means 21 First storage part 23 Interlocking control part (interlocking control part)
25 (25a) Operator A (operating means)
25 (25b) Operator B (interlocking operation means)
27 (27a) Display A (display means)
27 (27b) Display B (display means)
29 Analyzer (analysis means)
31 Database for analysis 33 Second storage section (storage means)
35 Third storage (control data storage means)

Claims (4)

A voice adjustment console that performs control of a sense of distance from a sound source to a listener and control to localize the direction of the sound source from the sound heard by the listener,
An operation means for inputting position information indicating the position of the sound source and inputting a trajectory along which the sound source moves;
Analyzing means for analyzing a control amount of the non-control means in which the sense of distance and the localization in the direction are respectively controlled from the trajectory input by the operation means;
Storage means for storing the controlled variable analyzed by the analysis means;
A voice adjustment console characterized by comprising: An audio adjustment console that adjusts at least one of volume and sound quality for each of a plurality of input audio signals,
Non-control means for adjusting at least one of the volume and the sound quality;
Control data storage means for storing control data for controlling the non-control means at an arbitrary time interval and an arbitrary control amount;
Interlocking operation means for designating control data stored in the control data storage means;
The control data designated by the interlock operation means is read at an arbitrary speed from the control data storage means, and the interlock control means for controlling the non-control means in conjunction with each other;
A voice adjustment console characterized by comprising:   It further comprises display means for displaying the positional relationship between the position of the listener and the position of the speaker that outputs the sound controlled by the non-control means, the control result by the interlock control means, and the locus of movement of the sound source. The voice adjustment console according to claim 2. Control of the sense of distance from the sound source to the listener and control to localize the direction of the sound source from the sound heard by the listener, the reflected sound reflected by the direct sound emitted from the sound source and the direct sound A voice adjustment console that mixes
An operation means for inputting position information indicating the position of the sound source and inputting a trajectory along which the sound source moves;
Distance attenuation control means for controlling the distance between the direct sound emitted from the sound source and the reflected sound reflected by the direct sound based on the position information;
Direction localization control means for performing control to localize the direction of the sound source based on the position information;
Reflected sound control means for generating reflected sound from direct sound emitted from the sound source;
Interlock control means for controlling the distance attenuation control means, the direction localization control means, and the reflected sound control means in conjunction with each other;
Output means for allocating and outputting the direct sound and the reflected sound controlled by the interlock control means to the plurality of speakers;
A voice adjustment console characterized by comprising:

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