JP2016149766A - Characteristic adjustment device for audio signal, characteristic adjustment program for audio signal, and characteristic adjustment method for audio signal - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide a characteristic adjustment device for audio signal, a characteristic adjustment program for audio signal, and a characteristic adjustment method for audio signal that facilitate upconversion of an audio signal.SOLUTION: A characteristic adjustment device for audio signal is configured to adjust characteristics of an audio signal when generating audio signals of a plurality of channels to be output from speakers by upconverting input audio signals of two channels, and includes: a delay control part 150 which imparts a delay time to a second audio signal other than a first audio signal output from a speaker located on a front side in a depth direction and by a display etc.; and an equalizer control part 140 which adjusts a gain and a frequency. The delay time is set based upon the sum of a first delay time for generating a simulation primary reflected wave of the first audio signal and a second delay time set to be longer for the second audio signal of a speaker arranged on a rear side. The gain and frequency are set to be lower for the second audio signal of the speaker arranged on the rear side.SELECTED DRAWING: Figure 10

Description

  The present invention relates to an audio signal characteristic adjustment device, an audio signal characteristic adjustment program, and an audio signal characteristic adjustment method.

  2. Description of the Related Art Conventionally, a speaker array device that reproduces sound of a multi-channel sound system by using a plurality of speaker units arranged around a display, and is a speaker unit for original signals that reproduces sound signals of channels arranged in front of a viewer A wavefront synthesis processing unit that converts an acoustic signal of a channel disposed in front of the viewer into a drive signal of a wavefront synthesis method, a low-pass filter that removes a high frequency component of the drive signal, and 2. Description of the Related Art There is a speaker array device that includes a wavefront synthesis speaker unit that reproduces a drive signal from which high frequencies have been removed (see, for example, Patent Document 1).

JP 2014-204362 A

  For example, as disclosed in Patent Document 1, there is a speaker array device that reproduces a multi-channel acoustic signal (multi-channel audio signal).

  By the way, for example, when up-converting a 2-channel audio signal to create a multi-channel audio signal, the characteristics of the audio signal such as gain and frequency are manually adjusted according to the content and type of the video and audio signal. It was set in.

  It is desirable to change the setting of such acoustic signal characteristics depending on, for example, a drama scene or the content of a program.

  For this reason, for example, if the characteristics of the acoustic signal are manually set for each scene of the drama, an enormous amount of work is required, and the up-conversion of the audio signal cannot be easily performed. Moreover, since it takes time and labor, it is difficult to cope with live broadcasting.

  Accordingly, an object of the present invention is to provide an audio signal characteristic adjustment device, an audio signal characteristic adjustment program, and an audio signal characteristic adjustment method that facilitate the up-conversion of an audio signal.

  The audio signal characteristic adjusting apparatus according to the embodiment of the present invention converts a plurality of channels of audio signals output from a plurality of speakers arranged along the depth direction with respect to a display, a screen, or a stage into two-channel input audio signals. Is an audio signal characteristic adjusting device that adjusts the characteristics of the audio signals of the plurality of channels when the signal is up-converted and generated, on the near side in the depth direction of the audio signals of the plurality of channels. A first adjustment unit that gives a delay time to a second audio signal other than the first audio signal output from a speaker located beside the display, the screen, or the stage; and gains of audio signals of the plurality of channels And a second adjusting unit that adjusts the frequency, and the delay time generates a simulated primary reflected wave of the first audio signal. Is set based on the sum of a first delay time for the second delay time set to be longer for the second audio signal output from the speaker arranged behind the near side in the depth direction. The gain and the frequency are set to be lower as the second audio signal is output from a speaker disposed on the rear side in the depth direction.

  It is possible to provide an audio signal characteristic adjustment device, an audio signal characteristic adjustment program, and an audio signal characteristic adjustment method that facilitate the up-conversion of an audio signal.

1 is a perspective view of a computer system to which an audio signal characteristic adjusting device according to an embodiment is applied. FIG. 25 is a block diagram illustrating a configuration of a main part in a main body 511 of a computer system 510. It is an external view which shows the characteristic adjustment apparatus 100A of the audio | voice signal of embodiment. It is an external view which shows the characteristic adjustment apparatus 100A of the audio | voice signal of embodiment. It is an external view which shows the characteristic adjustment apparatus 100A of the audio | voice signal of embodiment. It is an external view which shows the characteristic adjustment apparatus 100A of the audio | voice signal of embodiment. It is a figure which shows the sound system of a 22.2 channel. It is a figure which shows the 22.2 channel acoustic system of another form. It is a figure which shows the circuit structure of the up converter 1 which connects the characteristic adjustment apparatus 100 of the audio | voice signal of embodiment. It is a figure showing the functional block contained in the characteristic adjustment apparatuses 100 and 100A of an audio | voice signal. It is a figure which shows the data structure of parameter data. It is a figure which shows arrangement | positioning of a 5.1 channel and 7.1 channel speaker. It is a figure which shows arrangement | positioning of a 5.1 channel and 7.1 channel speaker. It is a figure which shows the circuit structure of the up-converter 1A of the 1st modification of embodiment. It is a figure which shows the GUI screen which selects the parameter data in the 1st modification of embodiment. It is a block diagram which shows the structure of the characteristic adjustment apparatus 100C of the audio | voice signal by the 2nd modification of embodiment.

  DESCRIPTION OF THE PREFERRED EMBODIMENTS Embodiments to which an audio signal characteristic adjusting device, an audio signal characteristic adjusting program, and an audio signal characteristic adjusting method of the present invention are applied will be described below.

<Embodiment>
FIG. 1 is a perspective view showing an audio signal characteristic adjusting apparatus 100 according to an embodiment. The audio signal characteristic adjusting apparatus 100 includes a computer system 510 and an up-converter 1. A computer system 510 shown in FIG. 1 includes a main body 511, a display 512, a keyboard 513, a mouse 514, and a modem 515. The up converter 1 is connected to the computer system 510 by a cable 1C. The computer system 510 is, for example, a PC (Personal Computer).

  The up-converter 1 of the audio signal characteristic adjustment device 100 is a device that performs a delay process that gives a delay time to the audio signal and an adjustment process that adjusts the gain and frequency of the audio signal. The computer system 510 functions as a setting device (setting device for the up-converter 1) that sets the delay time, gain, and frequency in the up-converter 1.

  The main body 511 includes a CPU (Central Processing Unit), an HDD (Hard Disk Drive), a disk drive, and the like. The display 512 is a display unit that displays an analysis result or the like on the display screen 512A according to an instruction from the main body unit 511, and may be a liquid crystal monitor, for example. The keyboard 513 is an input unit for inputting various information to the computer system 510. The mouse 514 is an input unit that designates an arbitrary position on the display screen 512 </ b> A of the display 512. The modem 515 accesses an external database or the like and downloads a program or the like stored in another computer system.

  A program for causing the computer system 510 to function as a setting device for the up-converter 1 is stored in a portable recording medium such as a disk 517 or a recording medium 516 of another computer system using a communication device such as a modem 515. Downloaded to the computer system 510 and compiled.

  A program for causing the computer system 510 to have a function as a setting device for the upconverter 1 causes the computer system 510 to operate as the setting device for the upconverter 1. This program may be stored in a computer-readable recording medium such as the disk 517, for example. The computer-readable recording medium is limited to a portable recording medium such as a disk 517, an IC card memory, a magnetic disk such as a floppy (registered trademark) disk, a magneto-optical disk, a CD-ROM, or a USB (Universal Serial Bus) memory. It is not something. The computer-readable recording medium includes various recording media that can be accessed by a computer system connected via a modem 515 or a communication device such as a LAN.

  FIG. 2 is a block diagram illustrating a configuration of a main part in the main body 511 of the computer system 510. The main body 511 includes a CPU 521 connected via a bus 520, a memory 522 including a RAM or a ROM, a disk drive 523 for the disk 517, and a hard disk drive (HDD) 524. In the embodiment, the display 512, the keyboard 513, and the mouse 514 are connected to the CPU 521 via the bus 520, but these may be directly connected to the CPU 521. The display 512 may be connected to the CPU 521 via a known graphic interface (not shown) that processes input / output image data.

  In the computer system 510, a keyboard 513 and a mouse 514 are input units of the setting device for the up-converter 1. The display 512 is a display unit that displays a setting result or the like by the setting device of the up-converter 1 on the display screen 512A.

  The computer system 510 is not limited to the configuration shown in FIGS. 1 and 2, and various well-known elements may be added or alternatively used.

  In FIG. 2, the computer system 510 having a function as a setting device for the up-converter 1 is used to set the delay time, gain, frequency, and the like in the up-converter 1. The up-converter 1 may hold data representing the delay time, gain, frequency, and the like.

  In this case, the up converter 1 may include a memory that stores data representing delay time, gain, frequency, and the like, and a processing unit that reads and sets data representing delay time, gain, frequency, and the like from the memory. That's fine.

  After the delay time, gain, frequency, etc. are set in the up-converter 1, the computer system 510 may be disconnected from the up-converter 1. Further, a memory in which data representing delay time, gain, frequency, and the like are stored in advance may be mounted in the upconverter 1 without connecting the computer system 510 to the upconverter 1.

  Next, the audio signal characteristic adjusting apparatus 100A according to the embodiment including the memory as described above will be described with reference to FIGS.

  3 to 6 are external views showing the sound signal characteristic adjusting apparatus 100A according to the embodiment. 3 to 6 show four surfaces (a plane, a front surface, a back surface, and a right side surface) of the sound signal characteristic adjusting device 100A.

  The sound signal characteristic adjustment device 100A includes a housing 101 and includes the up-converter 1 therein. As an example, the housing 101 is a so-called 1U (one-you) size metal or resin housing. A display unit 102 and operation units 103 and 104 are provided on the front surface of the housing 101. The display unit 102 is a liquid crystal panel or the like, and displays the volume, the level of each channel, and the like. The operation unit 103 includes switches for adjusting a master level (MASTER LEVEL), selecting an input (INPUT) and an output (OUTPUT), and the like. The operation unit 104 includes switches for selecting an entry key (ENTRY), a function (FUNCTION), a monitor (MONITOR), a channel (CH SELECT), and the like.

  On the rear surface of the housing 101, a power connector, a connector for inputting and outputting various signals, and the like are arranged. The plane (upper surface) of the housing 101 is sealed, and a cooling hole 105 is provided on the right side surface. The bottom surface of the housing 101 is sealed in the same manner as the flat surface (upper surface), and a cooling hole is provided on the left side surface in the same manner as the right side surface.

  FIG. 7 is a diagram showing a 22.2 channel sound system. The 22.2 channel sound system has been developed with the in mind to match 8K Super Hi-Vision video. As an example, the 22.2 channel acoustic system shown in FIG. 7 has a configuration in which speakers are arranged at vertices, sides, or points on a surface of a cube.

  In FIG. 7, description will be made using XYZ coordinates. The origin of the XYZ coordinates is the position of the speaker BtFL, and defines the X, Y, and Z axes as shown. The XY plane is parallel to the horizontal plane, and the Z axis is parallel to the vertical axis.

  The display 50 is installed on the YZ plane.

  The 22.2 channel acoustic system consists of 24 speakers LFE1, LFE2, BtFL, BtFC, BtFR, FL, FLc, FC, FRc, FR, SiL, SiR, BL, BC, BR, TpFL, TpFC, TpFR, TpSiL , TpC, TpSiR, TpBL, TpBC, TpBR.

  LFE1 and LFE2 correspond to 22.2 channel “.2” channels and are subwoofers.

  Speakers BtFL, BtFC, BtFR, FL, FLc, FC, FRc, FR, SiL, SiR, BL, BC, BR, TpFL, TpFC, TpFR, TpSiL, TpC, TpSiR, TpBL, TpBC, TpBR are 22.2 channels Corresponds to “22” channel.

  These 24 speakers are installed in accordance with a hall where an acoustic system is installed, a space such as a room, or a space for public viewing. Here, the X-axis direction is referred to as a depth direction such as a hall or a room, the side closer to the display 50 is referred to as front (F), and the side far from the display 50 is referred to as back (B). This is because when the viewer is at the ideal position at the center of the 24 speakers and views the display 50, the front-rear direction in the depth direction is based on whether the viewer is front (front) or rear (back). This is because of setting.

  The speakers LFE1 and LFE2 are speakers set on the left and right sides of the display 50 in a place where the front end (FE: Front End) is low (L: Low).

  The speakers BtFL, BtFC, and BtFR are positioned at the left (L), center (C), and right (R) of the lowest bottom (Bt: Bottom) on the front side, respectively. The positions of the speakers BtFL, BtFC, and BtFR in the Z-axis direction are the same height as the origin of the XYZ coordinate system.

  The speakers FL, FLc, FC, FRc, and FR are, at the center height in the Z-axis direction, on the front (F) side, left (L), left near the center (Lc), center (C), center, respectively. Located right (Rc) and right (R).

  The speakers SiL and SiR are located on the left (L) and right (R), respectively, at the center height in the Z-axis direction and at the center position of the depth in the X-axis direction. Si represents a side, which means that it is located on the lateral side of the viewer at an ideal position.

  The speakers BL, BC, and BR are positioned at the left (L), the center (C), and the right (R), respectively, on the back (B) side of the depth in the X-axis direction at the center height in the Z-axis direction. .

  Speakers TpFL, TpFC, and TpFR are positioned at the left (L), center (C), and right (R), respectively, on the front (F) side at the highest position (top: Tp (Top)) in the Z-axis direction. To do.

  The speakers TpSiL, TpC, and TpSiR have the left (L) and center (S) at the highest position (top: Tp (Top)) in the Z-axis direction and at the center position (side: Si) of the depth in the X-axis direction, respectively. C) Located on the right (R). The speaker TpC is located directly above the viewer at the ideal position.

  The speakers TpBL, TpBC, and TpBR are located at the highest position (top: Tp (Top)) in the Z-axis direction, and on the back (B) side of the depth in the X-axis direction, respectively, the left (L), the center (C), Located on the right (R).

  In the following, the speakers FL and FR are handled as channels 1 and 2, respectively. Also, speakers FC, LFE1, BL, BR, FLc, FRc, BC, LFE2, SiL, SiR, TpFL, TpFR, TpFC, TpC, TpBL, TpBR, TpSiL, TpSiR, TpBC, BtFC, BtFL, BtFR, respectively Treated as channels 3-24.

  FIG. 8 is a diagram showing another 22.2 channel sound system. FIG. 8 shows 24 speakers LFE1, LFE2, BtFL, BtFC, BtFR, FL, FLc, FC, FRc, FR, SiL, SiR, BL, BC, BR, TpFL, TpFC, TpFR, TpSiL, TpC, TpSiR , TpBL, TpBC, TpBR arrangement.

  In the 22.2 channel sound system shown in FIG. 8, the 24 speakers are arranged on a spherical surface centered on the virtual viewer's head. This is because all the 24 speakers are arranged at a position equidistant from the virtual viewer's head.

  Here, the height of 24 speakers (the height in the Z direction in FIG. 7) will be described in three stages.

  FIG. 8A shows middle-layer speakers FL, FLc, FC, FRc, FR, SiL, SiR, BL, BC, and BR. These are all arranged on one circumference.

  FIG. 8B shows top layer speakers TpFL, TpFC, TpFR, TpBL, TpBC, TpBR, TpSiL, TpC, and TpSiR. Of the nine speakers in the top layer, all but the speaker TpC are arranged on one circumference. The speaker TpC is disposed on the top of the spherical surface. In FIG. 8B, for convenience of explanation, eight speakers in the top layer other than the speaker TpC are shown on the circumference having substantially the same diameter as that in FIG. 8A. However, in the middle layer and the top layer, there are speakers. The diameter of the circumference placed can be different.

  FIG. 8C shows bottom layer speakers LFE1, LFE2, BtFL, BtFC, and BtFR. These are all arranged on one circumference. The diameter of the circumference where the five speakers in the bottom layer are arranged is smaller than the diameter of the circumference where the speakers in the middle layer and the top layer are arranged.

  FIG. 8D shows a cross-sectional view of the positional relationship between the top layer, the middle layer, and the bottom layer. Note that a certain allowable range is set for the arrangement of the 24 speakers. Detailed arrangement can be found in, for example, NHK STRL R & D / No.148 / 2014. 12-P. 21 “Development and Standardization Trend of 8K Super Hi-Vision Sound Production System”.

  FIG. 9 is a diagram illustrating a circuit configuration of the up-converter 1 according to the embodiment. Here, the up-converter 1 connected to the computer system 510 shown in FIG. 1 will be described.

  The up converter 1 includes an input terminal 2, an amplifier 3, a separator 4, a synchronization adjustment unit 5, an SRC (Sample Rate Converter) 6, a separator 7, a bus 8, an input terminal 9, a DSP (Digital Signal Processor) 10, and an encoder 20. , An output terminal 21, a matrix output unit 22, and a monitor output circuit 23.

  The input terminal 2 is a terminal to which a 2-channel audio signal is input. For example, a BNC plug connector may be used. A stereo audio signal is input from the sound source to the input terminal 2. Stereo audio signals are input to one input terminal 2 as AES-EBU standard digital signals. If the audio signal is monaural, the same audio signal for L (left) and R (right) may be used to obtain a 2-channel audio signal.

  The amplifier 3 amplifies and outputs a two-channel audio signal input to the input terminal 2.

  The separator 4 separates and outputs the audio signal amplified by the amplifier 3 to the synchronization adjustment unit 5 and the SRC 6.

  The synchronization adjusting unit 5 is an adjusting unit for synchronizing the video signal and the audio signal.

  The SRC 6 is a converter that converts a sampling rate. Here, the sampling rate is converted in order to convert a 2-channel audio signal into a 22.2 channel audio signal.

  The separator 7 separates the audio signal output from the SRC 6 into two audio signals for L (left) and R (right) and outputs them to the bus 8.

  The bus 8 is provided between the DSP 10 and the matrix output unit 22.

  The input terminal 9 is a terminal for connecting the computer system 510 (see FIG. 1). When the up-converter 1 is included in the audio signal characteristic adjusting apparatus 100A shown in FIGS. 3 to 6, an internal computer included in the characteristic adjusting apparatus 100A and having a memory may be connected to the input terminal 9.

  The DSP 10 includes 24 DSP units 10A. There are 24 DSP units 10A corresponding to 22.2 channels. The 24 DSP units 10A all have the same circuit configuration.

  Actually, 24 DSP units 10A for 1 channel to 24 channels are arranged in order from the upper side to the lower side in FIG. 9, but in FIG. 9, two DSPs for 1 channel and 2 channels are arranged. The section 10A is shown, and the DSP section 10A for 3 channels to 24 channels is not shown.

  Here, the internal configuration of the DSP unit 10A for one channel will be described. The DSP unit 10A includes mixers (MIX) 11L and 11R, a synthesizer 12, a level adjustment unit (LEVEL) 13, an equalizer (4band PEQ) 14, a delay unit (DELAY) 15, a switch 16, a master level adjustment unit (MASTER LEVEL). ) 17 and a mute switch (MUTE) 18.

  The mixing ratio setting units 11L and 11R adjust the ratio (ratio) for mixing two audio signals for L (left) and R (right), respectively. The ratio to be adjusted is set by a control signal MIX input from the computer system 510 connected to the input terminal 9 or the built-in computer.

  The synthesizer 12 mixes two audio signals for L (left) and R (right) output from the mixture ratio setting units 11L and 11R.

  The level adjustment unit 13 is a part that adjusts the signal level (volume) of the audio signal of the channel independently of the signal levels of the audio signals of other channels. The signal level is set by a control signal LEVEL input from the computer system 510 connected to the input terminal 9 or the built-in computer.

  The equalizer 14 divides the entire band into four bands, and sets a gain (GAIN), a frequency (FREQ), a gain type (TYPE), or Q for each band. These values are set to optimum values according to the contents and types of video and audio signals from a database in which values obtained by experiments are accumulated. The gain (GAIN), frequency (FREQ), gain type (TYPE), and Q are set by a control signal PEQ input from the computer system 510 connected to the input terminal 9 or the built-in computer.

  The delay unit 15 gives a delay time corresponding to the sum of the first delay time and the second delay time to the audio signal output from the speakers other than the speakers FL and FR. Therefore, the delay time given to the audio signal by the delay unit 15 of the DSP unit 10A of the channels 1 and 2 corresponding to the speakers FL and FR is 0 second.

  The speakers FL and FR of the channels 1 and 2 are speakers serving as a reference for the first delay time and the second delay time described below.

  The first delay time is used to generate a simulated primary reflected wave (simulated primary reflected wave) of the audio signal (first audio signal) output from the speakers (FL, FR) of channels 1 and 2. This time is set to be given to the audio signal (second audio signal) output from the 24 speakers.

  In addition, about the audio | voice signal (2nd audio | voice signal) output from the speaker of channel 3, 4, 7, 8, 10, 13-15 located in the front of a depth direction, without giving 1st delay time, You may set so that the audio | voice signal same as the audio | voice signal (1st audio | voice signal) output from the speaker (FL, FR) of the channels 1 and 2 may be output.

  The primary reflected wave has a higher signal level than the higher-order reflected wave higher than the secondary reflected wave and arrives the earliest, so it is dominant among the reflected waves excluding the direct sound that reaches the viewer directly without being reflected. Is. Therefore, the sound signal characteristic adjusting apparatuses 100 and 100A perform approximation using only the primary reflected wave, and simulate the sound signal (first sound signal) output from the speakers 1 and 2 of channels 1 and 2 (FL and FR). A primary reflected wave (simulated primary reflected wave) is generated for channels 3 to 24.

  When a human recognizes the direction of sound arrival, the judgment is first made by comparing the time difference between the sound reaching the left and right ears and the difference in volume. Next, the person can recognize the environment around him by listening to the primary reflection sound reflected from the objects around him. Experience to date by recognizing the time difference from the sound that arrived directly, the difference in sound quality caused by the absorption of the sound component, or the sound added by the resonance of the reflector To identify the environment around you. In what kind of environment the sound was emitted, where I was listening to this sound, and what kind of environment it was in the middle of propagation, It can be recognized. After that, information on reverberation (high-order reflection) remaining in the space is added, and people can understand the details of the surrounding environment.

  In the signal processing performed here, the high-order reflection processing is complicated by many conditions, and the influence of the primary reflected sound is large before adding the influence of the high-order reflection. ing.

  In addition, when thinking about music up-conversion, if the music already completed in stereo is expanded to a multi-surround space, up-conversion may greatly change the original world of music. Therefore, it is necessary to pay close attention. As an example, the up-converters that have been made so far have a strong reverberant sound after processing. Not adding higher-order reflections more than necessary helps to reduce the reverberation.

  The audio signal characteristic adjusting apparatuses 100 and 100A according to the embodiments are not intended to simulate listening to stereo sound reproduced from two pairs of speakers placed on the hall stage at the hall seat. The goal is to reproduce the sound field that is completed in stereo with a multi-channel sound system with the utmost emphasis on not changing the world view.

  By using the characteristics of human sound source and sound field environment recognition, stereo sound sources (including monaural sound sources) can be played back in a multi-channel spatial sound system without increasing the reverberation and reverberation more than necessary. Made for purpose.

  Therefore, the sound signal characteristic adjusting apparatuses 100 and 100A perform approximation using only the primary reflected wave, and simulate the sound signal (first sound signal) output from the speakers 1 and 2 of channels 1 and 2 (FL and FR). A primary reflected wave (simulated primary reflected wave) is generated for channels 2 to 24.

  The second delay time is determined by the speakers (BL, BR, BC, SiL, SiR, TpC, Channels 5, 6, 9, 11, 12, 16-22 located at the center (side) and back in the depth direction. This is a delay time given to an audio signal (second audio signal) output from TpBL, TpBR, TpSiL, TpSiR, TpBC, BtFC).

  The second delay time corresponds to the position of the speakers (FL, FR) of channels 1 and 2, and the speakers of channels 5, 6, 9, 11, 12, 16 to 22 (BL, BR, BC, SiL, SiR, TpC, In order to make the viewer feel that there is a sound source on the front side where the display 50 is present in consideration of the position difference of TpBL, TpBR, TpSiL, TpSiR, TpBC, BtFC), channels 5, 6, 9, 11 12 and 16 to 22 are set to be given to the audio signal (second audio signal) output from the speakers (BL, BR, BC, SiL, SiR, TpC, TpBL, TpBR, TpSiL, TpSiR, TpBC, BtFC). It is time to be. For this reason, the second delay time is set longer as it goes to the back side than the front side.

  The second delay time is output by the speaker located behind (back) the speakers FL and FR with respect to the audio signal output by the speakers FL and FR located in the foremost (frontmost) position in the depth direction of the hall. This is the time to delay the audio signal.

  By delaying the audio signal emitted by the object or living thing of the image displayed on the display 50 in front of the viewer on the back side from the front side, the audio signal is received from the front side. This is to allow the viewer to audibly recognize what is being heard.

  It is output from speakers (FC, LFE1, FLc, FRc, LFE2, TpFL, TpFR, TpFC, BtFL, BtFR) of channels 3, 4, 7, 8, 10, 13 to 15 located at the front in the depth direction. The audio signal may be given a delay time based on the difference from the position of the speakers (FL, FR) of the channels 1 and 2.

  The delay unit 15 gives the audio signal a delay time set based on the sum of the first delay time and the second delay time. In addition to the case where the sum of the first delay time and the second delay time is set as the delay time, for example, the sum obtained after multiplying the first delay time and / or the second delay time by a coefficient is delayed. This means that the time may be set, or the sum obtained by multiplying the sum of the first delay time and the second delay time by a coefficient may be set as the delay time.

  The coefficient is, for example, the distance between the front (front) speaker and the rear (back) speaker relative to the front (front) speaker, or the front (front) speaker and the front (front). It may be set according to the acoustic characteristics of the hall between the speaker on the back side and the speaker on the back side.

  The coefficient is a positive value, and no matter what value the coefficient is set to, the audio signal output from the front (front) speakers FL, FR is behind (back) the speakers FL, FR. The audio signal output from the speaker on the) side is delayed.

  The delay time is set by a control signal DELAY input from the computer system 510 connected to the input terminal 9 or the built-in computer.

  The switch 16 is a switch for switching on / off of each channel. The switch 16 may be configured to be switched on / off by the computer system 510 or the built-in computer.

  The master level adjustment unit 17 is an adjustment unit for adjusting the volume of the entire sound system, and is set to the same level in all channels. Note that the master level adjustment unit 17 may be configured to be switched on / off by the computer system 510 or the built-in computer.

  The mute switch (MUTE) 18 is a switch for muting the volume of the entire sound system. The mute switch (MUTE) 18 may be configured to be switched on / off by the computer system 510 or the built-in computer.

  The encoder 20 is an encoder in the MADI ENCODE format, receives an audio signal output from the 24-channel DSP unit 10A, and outputs the audio signal as one audio signal from the output terminal OUT to the output terminal 21.

  The output terminal 21 is a terminal that outputs an audio signal that is finally output by the acoustic system illustrated in FIG. 9. For example, a BNC plug connector may be used.

  The matrix output unit 22 is an output unit used when, for example, six channels are selected from 24 channels and 5.1 channel audio signals are output.

  The monitor output circuit 23 is a circuit used when performing a noise check on each channel.

  FIG. 10 is a diagram illustrating functional blocks included in the sound signal characteristic adjusting apparatuses 100 and 100A.

  FIG. 10A shows functional blocks realized by the setting device 550 of the up-converter 1 and realized by the computer system 510 included in the audio signal characteristic adjusting device 100.

  The setting device 550 includes a main control unit 110, a mixing ratio control unit 120, a level control unit 130, an equalizer control unit 140, a delay control unit 150, and a database 160.

  The main control unit 110 supervises the processing of the setting device 550. The main control unit 110 performs on / off switching control of the switch 16 for each channel, volume control by the master level adjustment unit 17 for each channel, and mute control by the mute switch (MUTE) 18 for each channel. .

  The mixing ratio control unit 120 uses the parameter data stored in the database 160 to set a mixing ratio setting in order to set a ratio (ratio) for mixing two audio signals for L (left) and R (right). The control signal MIX is output to the units 11L and 11R.

  The level control unit 130 outputs a control signal LEVEL in order to control the signal level (volume) of the audio signal by the level adjustment unit 13 using the parameter data stored in the database 160.

  The equalizer controller 140 uses the parameter data stored in the database 160 to generate a control signal PEQ for setting the gain (GAIN), frequency (FREQ), gain type (TYPE), or Q for each channel. Output. The equalizer control unit 140 is an example of a second adjustment unit.

  The delay control unit 150 uses the parameter data stored in the database 160 to output a control signal DELAY for setting the delay time in the delay unit 15. The delay control unit 150 is an example of a first adjustment unit.

  The database 160 stores parameter data. The parameter data is for controlling the signal level (volume) of the control signal MIX for setting the ratio (ratio) for mixing two audio signals for L (left) and R (right) for each channel. Table format including values of control signal LEVEL, gain (GAIN), frequency (FREQ), gain type (TYPE), control signal PEQ for setting Q, and control signal DELAY for setting delay time It is data. The parameter data will be described with reference to FIG.

  FIG. 10B shows functional blocks realized by the built-in computer 100B of the sound signal characteristic adjusting apparatus 100A. The sound signal characteristic adjusting device 100A includes a built-in computer 100B and the up-converter 1.

  The built-in computer 100B includes a main control unit 110, a mixing ratio control unit 120, a level control unit 130, an equalizer control unit 140, a delay control unit 150, and a database 160. The configuration of the built-in computer 100B is the same as that of the setting device 550 shown in FIG. The upconverter 1 is connected to the built-in computer 100B. The database 160 of the built-in computer 100B stores data representing delay time, gain, frequency, and the like in advance.

  FIG. 11 is a diagram showing a GUI screen for selecting parameter data. Such a GUI screen is displayed on the display 512 of the computer system 510 (see FIG. 1) functioning as the audio signal characteristic adjusting apparatus 100.

  There are a plurality of parameter data according to the contents and types of video and audio signals, and the user of the audio signal characteristic adjusting apparatus 100 can select them. FIG. 11 shows one of a plurality of parameter data.

  FIG. 11 shows the data structure of the parameter data in order to show the parameter data for each of the channels 1 to 24.

  The parameter data is data in a table format in which the values of the control signals MIX, LEVEL, GAIN, FREQ, TYPE, Q, and DELAY of each channel from channel 1 to channel 24 are associated.

  Of these, the four values GAIN, FREQ, TYPE, and Q are set for each band by dividing the entire band into four bands. In FIG. 11, four bands are shown from the upper row to the lower row from the higher frequency band to the lower frequency band. Here, since specific numerical values are not shown,... Are written instead of numerical values.

  TYPE is set to the highest band and the lowest band, and Q is set to the two middle bands. For this reason, three values GAIN, FREQ, and TYPE are set for the highest band and the lowest band, and three values GAIN, FREQ, and Q are set for the two middle bands.

  GAIN, FREQ, TYPE, and Q are input to the up converter 1 (see FIG. 9) as the control signal PEQ.

  MIX represents a ratio (ratio) at which the mixture ratio setting units 11L and 11R output two audio signals for L (left) and R (right). The two audio signals for L (left) and R (right) output from the mixing ratio setting units 11L and 11R are synthesized (mixed) by the synthesizer 12.

  LEVEL represents the signal level (volume) of the audio signal controlled by the level adjusting unit 13.

  GAIN and FREQ represent the gain and frequency of the audio signal controlled by the equalizer control unit 140, respectively.

  The gain type (TYPE) represents whether the gain controlled by the equalizer control unit 140 is increased with respect to the increase in frequency or is decreased with respect to the increase in frequency.

  Q is a quality factor (representing the sharpness of the resonance peak of the resonant circuit. Generally, Q is expressed as Q = ω0 / ω2? Ω1, and ω2? Ω1 is called a half width. The resonance frequency ω0 at which the amplitude reaches its peak is a frequency that is half of the amplitude value of the resonance frequency ω0, and the higher frequency is ω2 and the frequency that is equal (ω2−ω0) smaller than ω0 is ω1.

  DELAY represents a delay time that the delay unit 15 gives to the audio signal. The delay time represented by DELAY is set based on the sum of the first delay time and the second delay time described above.

  Since there are a plurality of parameter data as described above in accordance with the contents and types of video and audio signals, the user of the audio signal characteristic adjusting device 100 selects one of them to display 512 (see FIG. 1). ) As shown in FIG.

  The values of the control signals MIX, LEVEL, GAIN, FREQ, TYPE, Q, and DELAY of each channel from 1 channel to 24 channels included in the parameter data are optimum values obtained through experiments, and may be used as they are. The value may be increased or decreased by using an upward triangle button and a downward triangle button on the right side of each numerical value on the GUI screen.

  As shown in FIG. 11, in the parameter data, FREQ, GAIN, and Q are adjusted such that the longer the speaker position is, the longer the delay time and the lower the high frequency. With such a setting, it is possible to make the sound audible from the front to the viewer in the reproduction sound environment of 24 speakers.

  Since there is the display 50 in front of the viewer, the viewer can hear as if sound is coming from the video displayed on the front display 50. By using such parameter data, it is possible to maximize the presence of sound and video.

  Note that in the audio signal characteristic adjusting apparatus 100A shown in FIGS. 3 to 6, data representing delay time, gain, frequency, and the like stored in advance in the database 160 of the built-in computer 100B is used. And the frequency are set.

  As described above, according to the audio signal characteristic adjusting apparatuses 100 and 100A of the embodiment, when the 2-channel audio signal is up-converted to the 22.2 channel, the audio signal of the channels 1 and 2 is converted. By generating an audio signal having a delay time corresponding to a simulated primary reflected wave for channels 3 to 24, a 22.2 channel acoustic system can be used without using a higher order reflected wave higher than the secondary reflected wave. Can be obtained.

  In addition, in the up-conversion, in addition to the delay time described above, the speaker channel located on the rear side is set so that the signal level in the high frequency range of the gain and frequency is lower, so that the presence of sound and video is maximized. To the limit.

  In addition, the speaker installed at the upper side drops and limits the low frequency component. The loudspeaker placed below the front limits the high frequency range. These are proved from the experimental results. Note that “upper” means an upper side of the center in the height direction among a plurality of speakers arranged in the height direction. In this case, a speaker located at the center in the height direction may be included in the upper speaker. The term “lower” refers to the lower side of the center in the height direction among a plurality of speakers arranged in the height direction. In this case, a speaker located at the center in the height direction may be included in the lower speaker. Moreover, the front means the front side rather than the center of the front-back direction among the speakers arrange | positioned in the front-back direction in the depth direction. In this case, a speaker located at the center in the front-rear direction may be included in the front speaker.

  In addition, an audio signal capable of sufficiently supporting a 22.2 channel acoustic system can be obtained without using a higher-order reflected wave equal to or higher than the second-order reflected wave, so that the apparatus configuration is very simple and low-cost. Thus, it is possible to provide the sound signal characteristic adjusting apparatuses 100 and 100A.

  In particular, by adding a sampling reverb or an existing reverb to the results obtained by the sound signal characteristic adjusting apparatuses 100 and 100A of the embodiment, various acoustic expressions can be easily and in detail.

  Conventionally, since the characteristics of the acoustic signal such as gain and frequency are manually set according to the content and type of the video and the acoustic signal, the up-conversion of the audio signal cannot be easily performed. Therefore, it was difficult to support live broadcasting.

  In contrast, the sound signal characteristic adjusting apparatuses 100 and 100A according to the embodiment can instantaneously and easily use the selected parameter data by selecting the parameter data according to the content and type of the video and audio signals. Audio signals can be up-converted. For this reason, it does not take time and can respond to live broadcasting.

  Further, as described above, the sound signal characteristic adjusting devices 100 and 100A have a very simple device configuration. The up-converter 1 can also be reduced in size, and the audio signal characteristic adjusting device 100A is reduced in size so as to fit in a so-called 1U size. For this reason, for example, it can be mounted on a relay car and provided with 22.2 channels in a live broadcast program such as Ekiden to match the runner's voice with the 8K Super Hi-Vision video. In addition, since the sound signal characteristic adjusting device 100A stores data representing delay time, gain, frequency, and the like in the database 160 of the built-in computer 100B in advance, the sound signal can be easily adjusted without being set by the setting device 550. The signal can be upconverted.

  In the above description, the mode in which the 2-channel audio signal is up-converted to 22.2 channel has been described. However, the channel to be up-converted is not limited to 22.2 channel. For example, 5.1 channel, 7 channel .1 channel, 9.1 channel, 10.2 channel, 19.1 channel, etc., or more channels.

  12 and 13 are diagrams showing the arrangement of 5.1-channel and 7.1-channel speakers. In FIG. 12 and FIG. 13, the subwoofer is omitted. The subwoofer may be disposed near the center speaker C.

  In the 5.1 channel shown in FIG. 12, the speaker C is arranged in front of the viewer (front), and the speakers L and R are arranged on the left and right. Speakers Ls and Rs are arranged on the left and right sides slightly behind the viewer.

  The 7.1 channel shown in FIG. 13 has a configuration in which rear (back) speakers Lb and Rb are added to the 5.1 channel.

  Regarding the 5.1 channel arrangement, for example, there is a standard called ITU recommendation ITU-R BS.775-1, and for the 7.1 channel arrangement, there is an arrangement recommended by Dolby, for example.

  When the 5.1 channel audio signal is up-converted to 22.2 channel by the audio signal characteristic adjusting apparatus 100 or 100A, basically, the 2 channel audio signal is up-converted to 22.2 channel. Similarly, the audio signals output from the front left and right speakers L and R are handled as the first audio signal.

  The first delay time in the case of 5.1 channel is set so as to be given to the audio signal output from the speakers of channels 3 to 24 and output from the speakers of channels 3 to 24 as in the case of 2 channels. What is necessary is just to give to the audio | voice signal to be performed.

  In addition, the second delay time in the case of 5.1 channel is the same as that in the case of 2 channel, and the speakers of channels 5, 6, 9, 11, 12, 16 to 22 (BL, BR, BC, SiL, SiR, What is necessary is just to give to the audio | voice signal output from TpC, TpBL, TpBR, TpSiL, TpSiR, TpBC, BtFC).

  At this time, the 5.1 channel speakers Ls and Rs and the 22.2 channel speakers SiL, SiR, BL, BR, TpSiL, TpSiR, TpBL, and TpBR are positioned in the 5.1 channel. The sound signal output from the speakers relatively close to the speakers Ls and Rs may be added with the characteristics of the sound signals output from the 5.1-channel speakers Ls and Rs.

  For example, for the audio signal output from the 22.2 channel speaker BL, the first delay time is given to the audio signal output from the speakers FL and FR. Further, a predetermined ratio of the delay time for the 5.1 channel speakers L and R and the second delay time of the 5.1 channel speaker Ls to the audio signal given the first delay time. The delay time synthesized in (1) may be given.

  Further, when the equalizer 14 sets the gain and frequency of the audio signal output from the 22.2 channel speaker BL, the gain and frequency are set with reference to the audio signal gain and frequency of the 5.1 channel speaker Ls. May be set.

  When the 7.1 channel audio signal is up-converted to 22.2 channel by the audio signal characteristic adjusting apparatus 100 or 100A, basically, the 2 channel audio signal is up-converted to 22.2 channel. In the same manner as the above, the audio signal output from the left and right speakers L, R on the front is handled as the first audio signal.

  The first delay time in the case of 7.1 channel is set so as to be given to the audio signal output from the speakers of channels 3 to 24 and output from the speakers of channels 3 to 24, as in the case of 2 channels. What is necessary is just to give to the audio | voice signal to be performed.

  In addition, the second delay time in the case of 7.1 channel is the same as that in the case of 2 channel, and the speakers (BL, BR, BC, SiL, SiR, Channels 5, 6, 9, 11, 12, 16-22) What is necessary is just to give to the audio | voice signal output from TpC, TpBL, TpBR, TpSiL, TpSiR, TpBC, BtFC).

  At this time, the positions in the halls are different from the 7.1 channel speakers Ls, Rs, Lb, Rb and 22.2 channel speakers SiL, SiR, BL, BR, TpSiL, TpSiR, TpBL, TpBR. For audio signals output from speakers relatively close to 7.1-channel speakers Ls, Rs, Lb, and Rb, the characteristics of audio signals output from 7.1-channel speakers Ls, Rs, Lb, and Rb are taken into account. May be.

  For example, for the audio signal output from the 22.2 channel speaker BL, the first delay time is given to the audio signal output from the speakers FL and FR. Further, in addition to the audio signal given the first delay time, the delay time for the 7.1-channel speakers L and R of the audio signal obtained by averaging the audio signals of the 7.1-channel speakers Ls and Lb; A delay time obtained by combining the second delay time with a predetermined ratio may be given.

  In addition, when setting the gain and frequency of the audio signal output from the 22.2 channel speaker BL by the equalizer 14, the gain and frequency of the audio signal of the 7.1 channel speakers Ls and Lb are referred to as a gain. And the frequency may be set.

  As described above, the 5.1 channel or 7.1 channel audio signal can be easily and simply up-converted to 22.2 channel.

  FIG. 14 is a diagram illustrating a circuit configuration of an upconverter 1A according to a first modification of the embodiment. The up-converter 1A has a configuration capable of up-converting either the audio signals of the 2-channel speakers L and R and the audio signals of the 5.1-channel speakers Ls, L, C, R, Rs, and LFE.

  Here, signals output from 5.1 channel speakers L and R are handled as channels 1 and 2, respectively. Further, as an example, a signal output from the speaker C is handled as channel 3, a signal output from the subwoofer LFE is handled as channel 4, and signals output from the speakers Ls and Rs are handled as channels 5 and 6.

  Further, regarding the 24 DSP units 10A for 1 channel to 24 channels, the inside of the 1 channel DSP unit 10A is shown in detail, and the illustration of the DSP unit 10A for 2 channels to 24 channels is omitted.

  The up-converter 1 has input terminals 2A, 2B, 2C, amplifiers 3A, 3B, 3C, separators 4A, 4B, 4C, synchronization adjustment units 5A, 5B, 5C, SRC (Sample Rate Converter) 6A, 6B, 6C, and separation. 7A, 7B, 7C, bus 8, input terminal 9, DSP (Digital Signal Processor) 10, encoder 20, output terminal 21, matrix output units 22A, 22B, 22C, and monitor output circuit 23.

  The up-converter 1A includes an input terminal 2, an amplifier 3, a separator 4, a synchronization adjustment unit 5, an SRC 6, a separator 7 and a matrix output unit 22 of the up-converter 1 for two-channel audio signals shown in FIG. 2A, amplifier 3A, separator 4A, synchronization adjustment unit 5A, SRC 6A, separator 7A, matrix output unit 22A, and the input terminal 2B for the 2-channel audio signal up-converter 1 shown in FIG. 2C, amplifiers 3B and 3C, separators 4B and 4C, synchronization adjustment units 5B and 5C, SRCs 6B and 6C, separators 7B and 7C, and matrix output units 22B and 22C.

  The input terminal 2A, the amplifier 3A, the separator 4A, the synchronization adjustment unit 5A, the SRC 6A, the separator 7A, and the matrix output unit 22A are for audio signals of 1 and 2 channels, and the input terminal 2B, the amplifier 3B, the separator 4B, The synchronization adjustment unit 5B, SRC 6B, separator 7B, and matrix output unit 22B are for audio signals of 3 and 4 channels. Input terminal 2C, amplifier 3C, separator 4C, synchronization adjustment unit 5C, SRC 6C, separator 7C, The matrix output unit 22C is for audio signals of 3 and 4 channels.

  Input terminals 2B and 2C, amplifiers 3B and 3C, separators 4B and 4C, synchronization adjustment units 5B and 5C, SRC6B and 6C, separators 7B and 7C, matrix output units 22B and 22C, input terminal 2A, amplifier 3A, and separation 4A, synchronization adjustment unit 5A, SRC 6A, separator 7A, and matrix output unit 22A are provided in parallel.

  Since the input terminal 2A, the amplifier 3A, the separator 4A, the synchronization adjustment unit 5A, the SRC 6A, the separator 7A, and the matrix output unit 22A are for audio signals of 1 and 2 channels, respectively, the input terminal 2 shown in FIG. This is the same as the amplifier 3, separator 4, synchronization adjusting unit 5, SRC 6, separator 7, and matrix output unit 22.

  The audio signals of 3, 4, 5, and 6 channels processed by the input terminals 2B and 2C, the amplifiers 3B and 3C, the separators 4B and 4C, the synchronization adjustment units 5B and 5C, the SRCs 6B and 6C, and the separators 7B and 7C are What is necessary is just to use for the adjustment of the audio | voice signal output from the speaker of channels 3-24.

  Each of the 24 DSP units 10A has a configuration different from that of the 2-channel DSP unit 10A (see FIG. 9) in accordance with 5.1 channels. Hereinafter, differences will be described.

  The DSP unit 10A is wired so that 5.1-channel speakers Ls, L, C, R, Rs, and LFE are input in addition to the audio signals of the 2-channel speakers L and R. The LFE is a 5.1 channel subwoofer.

  The DSP unit 10A includes switches 11A and 11B, mixers (MIX) 11C and 11D, a synthesizer 12, a switch 12A, a level adjustment unit (LEVEL) 13, an equalizer (4band PEQ) 14, a delay unit (DELAY) 15, and a switch 16 A master level adjustment unit (MASTER LEVEL) 17 and a mute switch (MUTE) 18.

  The switches 11A and 11B select one of the switches 11A and 11B, whether the input signal is from an AES digital signal terminal or input from a MADI signal terminal.

  The switches 11A and 11B can select one of the audio signals of the 2-channel speakers L and R and the 5.1-channel speakers Ls, L, C, R, and Rs. The case is the same as the DSP unit 10A shown in FIG. The reason why the MADI input can be selected by the switches 11A and 11B is to enable processing of a surround signal having the number of channels of 5.1ch or more. When two channels are selected, the processing is the same as that of the DSP unit 10A shown in FIG.

  When selecting the 5.1 channel speakers Ls, L, C, R, and Rs with the switches 11A and 11B, select any two of the five audio signals other than the LFE included in the 5.1 channel. can do.

  The mixing ratio setting units 11C and 11D adjust the ratio (ratio) for mixing two audio signals selected by the switches 11A and 11B, respectively. The ratio to be adjusted is set by a control signal MIX input from the computer system 510 connected to the input terminal 9 or the built-in computer 100C.

  The synthesizer 12 mixes the two audio signals output from the mixing ratio setting units 11C and 11D.

  The switch 12A is a switch for selecting the sound of the 5.1-channel speaker LFE (subwoofer). When sending to LFEch, an equivalent signal is sent to LFE-L and LFE-R in parallel.

  The level adjustment unit 13, the equalizer 14, the delay unit 15, the switch 16, the master level adjustment unit 17, the mute switch 18, the encoder 20, and the output terminal 21 are the same as the DSP unit 10A illustrated in FIG.

  FIG. 15 is a diagram showing a GUI screen for selecting parameter data in the first modification of the embodiment. Such a GUI screen is displayed on the display 512 of the computer system 510 (see FIG. 1).

  There are a plurality of parameter data according to the contents and types of video and audio signals, and the user of the audio signal characteristic adjusting apparatus 100 can select them. FIG. 15 shows one of a plurality of parameter data (L200).

  FIG. 15 shows control signals LEVEL, MIX, and C0 for each channel from channel 1 to channel 24.

  LEVEL represents the signal level (volume) of the audio signal controlled by the level adjusting unit 13.

  MIX represents a ratio (ratio) at which the mixture ratio setting units 11C and 11D output two audio signals. The two audio signals output from the mixing ratio setting units 11C and 11D are synthesized (mixed) by the synthesizer 12.

  C0 represents the amount of synthesis (mixing) of the audio signals of the 5.1 channel speaker C, and C0 indicates that the synthesis (mixing) is zero (0). As synthesis (mixing) increases, the value to the right of C increases. C0 is a control item for mixing 5.1ch front Lch and Rch audio signals. When C = 0 is displayed, L + R, that is, Lch and Rch signals are equally mixed. In addition to C0, the display examples include L-Max (Lch is set to the maximum value), L55 (Lch is 55%), L12 (Lch is 12%), R12 (Rch is 12%), R55 (Rch 55%) and R-Max (Rch is set to the maximum value).

  Using the GUI screen for selecting parameter data in the first modification of the embodiment, the audio signals of the 2-channel speakers L and R and the 5.1-channel speakers Ls, L, C, R, and Rs are used. , Any of the LFE audio signals can be up-converted.

  According to the modification of the embodiment, when the 5.1 channel audio signal is up-converted to 22.2 channel, the delay time corresponding to the simulated primary reflected wave of the channel 1 and 2 audio signals. Can be obtained for channels 3 to 24, thereby obtaining an audio signal that can sufficiently cope with a 22.2 channel acoustic system without using a higher-order reflected wave higher than the second-order reflected wave. it can.

  In the first modification of the embodiment, the 5.1 channel audio signal is up-converted to 22.2 channel. However, the audio signal that is the source of up-conversion is limited to 5.1 channel. For example, 7.1 channels may be used.

  Further, the channel number is not limited to 5.1 channel or 5.1 channel, and may be 5.2 channel, 7.2 channel, 7.4 channel, or the like having a larger number of sub-foors.

  In the second modification of the embodiment, a configuration in which the up-converter and the down-converter are combined into one device will be described with reference to FIG.

  FIG. 16 is a block diagram illustrating a configuration of an audio signal characteristic adjusting apparatus 100C according to a second modification of the embodiment.

  The sound signal characteristic adjusting apparatus 100C has a configuration in which a down converter is added to the sound signal characteristic adjusting apparatus 100A. More specifically, the audio signal characteristic adjustment device 100 </ b> C includes an up-converter 1 and a down-converter 200.

  The up-converter 1 receives a 2-channel audio signal (STEREO) or a 5.1-channel or 7.0-channel (5.1ch-) audio signal, and up-converts it to 22.2 channel for output.

  Further, the 22.2 channel audio signal is input to the up-converter 1, and the 22.2 channel audio signal is output as it is as the 22.2 channel audio signal and input to the down converter 200. The down-converter 200 down-converts the 22.2 channel audio signal into a 2-channel audio signal, or an audio signal of 5.1 channel or 7.0 channel (from 5.1ch) and outputs. As such a down converter 200, a known one may be used.

  Similarly to the sound signal characteristic adjusting device 100A, the sound signal characteristic adjusting device 100C can up-convert the sound signal to 22.2 channels and can down-convert the 22.2 channel sound signals by the down converter 200. it can. Such an audio signal characteristic adjusting apparatus 100C can be realized as a 1U (one-you) apparatus, for example, like the audio signal characteristic adjusting apparatus 100A shown in FIGS.

  Although the audio signal characteristic adjustment apparatus, the audio signal characteristic adjustment program, and the audio signal characteristic adjustment method according to the exemplary embodiments of the present invention have been described above, the present invention is specifically disclosed. The present invention is not limited to this embodiment, and various modifications and changes can be made without departing from the scope of the claims.

DESCRIPTION OF SYMBOLS 1 Up converter 2 Input terminal 3 Amplifier 4 Separator 5 Synchronization adjustment part 6 SRC
7 Separator 8 Bus 9 Input terminal 10 DSP
10A DSP unit 11L, 11R Mixing ratio setting unit 12 Synthesizer 13 Level adjustment unit 14 Equalizer 15 Delay unit 16 Switch 17 Master level adjustment unit 18 Mute switch 20 Encoder 21 Output terminal 22 Matrix output unit 23 Monitor output circuit 100, 100A Audio signal Characteristic adjustment device 110 main control unit 120 mixing ratio control unit 130 level control unit 140 equalizer control unit 150 delay control unit 160 database

Claims (6)

When the audio signals of a plurality of channels output from a plurality of speakers arranged along the depth direction and the width direction with respect to the display, screen, or stage are generated by up-converting the input audio signals of two channels, the plurality A sound signal characteristic adjusting device for adjusting the sound signal characteristics of the channels of
Among the audio signals of the plurality of channels, a delay time is added to a second audio signal other than the first audio signal output from a speaker located on the near side of the display, the screen, or the stage on the near side in the depth direction. A first adjustment unit for providing
A second adjustment unit for adjusting the gain and frequency of the audio signals of the plurality of channels,
The delay time includes a first delay time for generating a simulated primary reflected wave of the first audio signal, and the second audio signal output from a speaker arranged behind the near side in the depth direction. It is set based on the sum of the second delay time set as long as possible,
The audio signal characteristic adjusting apparatus, wherein the gain and the frequency are set to be lower as the second audio signal is output from a speaker disposed on the rear side in the depth direction.   The audio signal characteristic adjusting apparatus according to claim 1, wherein the second adjusting unit adjusts gains and frequencies of the audio signals of the plurality of channels for each band obtained by dividing the entire band into a plurality of bands.   3. The audio signal characteristic according to claim 1, wherein the second adjustment unit adjusts gains and frequencies of the audio signals of the plurality of channels such that the gain increases or decreases according to an increase in frequency. Adjustment device.   The second adjustment unit reduces the low frequency component of the audio signal of the channel corresponding to the speaker installed above, and reduces the high frequency of the audio signal of the channel corresponding to the speaker installed below the front. 4. The audio signal characteristic adjustment apparatus according to claim 1, wherein a gain and a frequency of the audio signals of the plurality of channels are adjusted. When the audio signals of a plurality of channels output from a plurality of speakers arranged along the depth direction and the width direction with respect to the display, screen, or stage are generated by up-converting the input audio signals of two channels, the plurality An audio signal characteristic adjustment program for adjusting the audio signal characteristics of
Computer
Among the audio signals of the plurality of channels, a delay time is added to a second audio signal other than the first audio signal output from a speaker located on the near side of the display, the screen, or the stage on the near side in the depth direction. And giving
Adjusting the gain and frequency of the audio signals of the plurality of channels, and
The delay time includes a first delay time for generating a simulated primary reflected wave of the first audio signal, and the second audio signal output from a speaker arranged behind the near side in the depth direction. It is set based on the sum of the second delay time set as long as possible,
The audio signal characteristic adjustment program, wherein the gain and the frequency are set to be lower as the second audio signal is output from a speaker arranged on the rear side in the depth direction. When the audio signals of a plurality of channels output from a plurality of speakers arranged along the depth direction and the width direction with respect to the display, screen, or stage are generated by up-converting the input audio signals of two channels, the plurality A method for adjusting the characteristics of an audio signal, which adjusts the characteristics of the audio signal of the channel,
Among the audio signals of the plurality of channels, the computer outputs second audio signals other than the first audio signal output from the display, the screen, or a speaker located beside the stage on the near side in the depth direction. Giving a delay time,
Adjusting the gains and frequencies of the audio signals of the plurality of channels, and the delay time includes a first delay time for generating a simulated primary reflected wave of the first audio signal, and a depth direction in the depth direction. It is set based on the sum of the second delay time which is set longer as the second audio signal output from the speaker arranged behind the near side,
The audio signal characteristic adjusting method, wherein the gain and the frequency are set to be lower as the second audio signal is output from a speaker disposed on the rear side in the depth direction.

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