JP2002176700A - Signal processing device and recording medium - Google Patents

Abstract

(57) [Summary] [PROBLEMS] To provide a signal processing device of an acoustic signal capable of responding to various demands of a viewer. SOLUTION: A signal processing device 1a processes an audio signal AS reproduced together with an image signal VS. Signal processing device 1
a, a memory 4 for storing a plurality of filter coefficients for correcting the audio signal AS, and a correction command for specifying a correction method for the audio signal AS received from outside the signal processing device 1a, and a memory based on the correction command. 4, a filter coefficient selection unit 3 for selecting at least one of the plurality of filter coefficients stored in the filter coefficient selection unit 3.
And a correction unit 5 that corrects the acoustic signal AS using at least one filter coefficient selected by (1).

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a signal processing apparatus for processing an audio signal reproduced together with an image signal, and more particularly, to provide a viewer with a sense of distance of a sound image adapted to the situation of the reproduced image signal. The present invention relates to a signal processing device that realizes a viewing environment in which sight and hearing match, and a recording medium that records such an image signal and an acoustic signal.

[0002]

2. Description of the Related Art In recent years, in addition to video as a recording medium capable of storing sound data and image data together, a laser disk (registered trademark) or a DVD (digital versatile disk) is used.
Optical discs such as discs have become very widespread, and AV (Audio / Vid) can be used in ordinary households using a laser disc player or a DVD player as a reproducing device.
eo) An environment where reproduction can be easily enjoyed is being prepared. In addition, there is a situation in which audio data and image data are compressed and converted into data according to the MPEG standard and the like, so that personal computers can enjoy more personal AV reproduction.

Generally, in these reproduction environments, matching between image data and acoustic data cannot be said to be sufficient. For example, an image changes from a near view to a distant view, or a sound image is fixed at a fixed position despite changing in the horizontal direction. Was left.

[0004] In order to solve this problem, various proposals have conventionally been made to improve the viewing situation enjoyed by viewers when reproducing image data and audio data.

For example, Japanese Patent Application Laid-Open No. 9-70094 discloses
Disclosed a technology that installs a sensor that detects the movement of the viewer's head and corrects the acoustic signal based on the output signal of the sensor, thereby changing the position of the sound image in accordance with the movement of the viewer's head. ing.

[0006] Further, International Publication WO95 / 22235.
Installed a sensor that detects the movement of the viewer ’s head,
A technique for performing sound source position control synchronized with an image based on an output signal of the sensor is disclosed.

[0007]

However, in the above-described conventional signal processing device, only a filter prepared in advance in the signal processing device can be used to correct the acoustic signal, and thus the viewer desires. It was not possible to correct the sound signal or reflect the intention of the content creator in the correction of the sound signal.

[0008] Even if a filter for realizing a desired audio signal correction by a viewer can be prepared in advance, an audio signal corrected by a personal computer or the like must be used. Complicated work was required in order to ensure matching.

Further, a plurality of methods have been proposed for signal processing for realizing movement of a sound image on a monitor plane for displaying an image, but a signal for realizing a sense of distance (a sense of depth) of a sound image with a small amount of memory and a small amount of calculation has been proposed. No treatment method has been proposed.

[0010] The present invention has been made in view of such conventional problems, and enables not only reproduction of image signals and audio signals but also response to various demands of viewers. , An audio signal processing device, and
It is an object of the present invention to provide a recording medium on which such an image signal and an acoustic signal are recorded.

[0011]

A signal processing apparatus according to the present invention is a signal processing apparatus for processing an audio signal reproduced together with an image signal, and stores a plurality of filter coefficients for correcting the audio signal. A memory and a correction command specifying a method of correcting the acoustic signal are received from outside the signal processing device, and at least one of the plurality of filter coefficients stored in the memory is selected based on the correction command. And a correction unit that corrects the acoustic signal using at least one filter coefficient selected by the filter coefficient selection unit.

With this configuration, it is possible to change the method of correcting the sound signal according to the change in the image signal or the change in the sound signal. Thereby, the viewer can hear sound suitable for the image displayed on the image display device from the speaker or the headphones. As a result, it is possible to prevent the viewer from feeling uncomfortable with the relationship between the sight and the hearing.

[0013] Further, according to this configuration, according to the acoustic characteristics of speakers or headphones used by the viewer and the acoustic characteristics based on physical individual differences such as the shape of the ears and faces of the viewer,
It is possible to change the method of correcting the acoustic signal. As a result, a better sound environment can be provided to the viewer.

Further, since the filter coefficients themselves are stored in the memory, the filter coefficients do not need to be received from outside the signal processing device during the reproduction of the image signal and the audio signal. Therefore, when the signal processing device receives the correction command, the filter coefficient can be finely switched according to the change in the image signal or the change in the acoustic signal. As a result, it is possible to change the method of correcting the audio signal in a manner that more reflects the intention of the creator of the image signal and the audio signal (content).

[0015] In one embodiment of the present invention, the correction command is input to the signal processing device by receiving a broadcast or a communication.

In one embodiment of the present invention, the correction command is recorded on a recording medium, and the correction command is input to the signal processing device by reproducing the recording medium.

With this configuration, it is possible to input a correction command to the signal processing device by reproducing the recording medium.

In one embodiment of the present invention, the memory receives at least one filter coefficient for correcting the acoustic signal from outside the signal processing device, and stores the received at least one filter coefficient in the memory. Adding to the plurality of filter coefficients stored in
Alternatively, it is configured such that the received at least one filter coefficient can be replaced with at least one of the plurality of filter coefficients stored in the memory.

With this configuration, the contents of the filter coefficients stored in the memory can be easily updated.

In one embodiment of the present invention, the at least one filter coefficient is recorded on a recording medium, and the at least one filter coefficient is input to the signal processing device by reproducing the recording medium. Is done.

With this configuration, it is possible to input at least one filter coefficient to the signal processing device by reproducing the recording medium.

In one embodiment of the present invention, the signal processing device further includes a buffer memory for temporarily storing the image signal and the audio signal, wherein the image signal and the audio signal are stored in the buffer memory. The speed at which the image signal and the acoustic signal are output from the buffer memory is faster than the speed at which the image signal and the acoustic signal are output from the buffer memory. The time required for the image signal and the audio signal to be output from the buffer memory is stored in the memory while the and the audio signal are output, and the at least one filter coefficient is stored in the memory. More than the time needed to be stored.

With this configuration, it is possible to reproduce the sound signal correction data recorded on the recording medium without interrupting the image signal and the sound signal output from the reproducing device.

In one embodiment of the present invention, the at least one selected filter coefficient is at least one representing a transfer function indicating an acoustic characteristic from a sound source to a viewer.
And a transfer function correction circuit for correcting a transfer function of the acoustic signal according to the at least one filter coefficient representing the transfer function.

According to this configuration, the viewer can perceive the virtual sound source using the speaker or the headphones.

In one embodiment of the present invention, the at least one selected filter coefficient includes at least one filter coefficient representing a transfer function indicating an acoustic characteristic of a direct sound from a sound source to a viewer; And at least one filter coefficient representing a reflected sound structure indicating the acoustic characteristics of the reflected sound reaching the viewer, and the correction unit,
A transfer function correction circuit for correcting a transfer function of the acoustic signal according to the at least one filter coefficient representing the transfer function; and a reflected sound for the acoustic signal according to the at least one filter coefficient representing the reflected sound structure. , And an adder that adds the output of the transfer function correction circuit and the output of the reflected sound addition circuit.

According to this configuration, it is possible to make the viewer perceive the virtual sound source with a small amount of calculation using the speaker or the headphones.

In one embodiment of the present invention, the at least one selected filter coefficient includes at least one filter coefficient representing a transfer function indicating an acoustic characteristic of a direct sound from a sound source to a viewer; And at least one filter coefficient representing a reflected sound structure indicating the acoustic characteristics of the reflected sound reaching the viewer, and the correction unit,
A transfer function correction circuit for correcting a transfer function of the acoustic signal according to the at least one filter coefficient expressing the transfer function; and a transfer function correction circuit for correcting the transfer function according to the at least one filter coefficient expressing the reflected sound structure. A reflected sound adding circuit for adding a reflected sound to the output.

According to this configuration, the virtual sound source can be more clearly perceived by the viewer with a small amount of calculation using a speaker or headphones.

In one embodiment of the present invention, the filter coefficient selection section automatically selects at least one of the plurality of filter coefficients stored in the memory based on the correction command. An automatic selection unit; and a manual selection unit that manually selects at least one of the plurality of filter coefficients stored in the memory.

With this configuration, the viewer can select whether the filter coefficient is automatically selected or manually selected.

In one embodiment of the present invention, the at least one filter coefficient representing the reflected sound structure is:
When the distance between the sound source and the viewer is a first distance, a first filter coefficient representing a reflected sound structure indicating acoustic characteristics of a reflected sound from the sound source to the viewer; A second filter coefficient expressing a reflected sound structure indicating an acoustic characteristic of a reflected sound from the sound source to the viewer when the distance to the viewer is a second distance different from the first distance.

With this configuration, it is possible to arbitrarily set the distance between the virtual sound source and the viewer.

In one embodiment of the present invention, the at least one filter coefficient representing the reflected sound structure is:
And a third filter coefficient expressing a reflected sound structure indicating acoustic characteristics of the reflected sound arriving at the viewer from a direction in a predetermined range.

With this configuration, it is possible to simulate the sound field desired by the viewer with higher accuracy.

In one embodiment of the present invention, the direction of the predetermined range is centered on a center of the viewer's head.
1 from the straight line connecting the sound source and the center of the viewer's head
This is a direction within a range forming an angle of 5 degrees or less.

With this configuration, it is possible to simulate the sound field desired by the viewer with higher accuracy.

In one embodiment of the present invention, the audio signal includes audio signals of a plurality of channels, and the filter coefficient selection unit selects a filter coefficient corresponding to each of the audio signals of the channels. .

With this configuration, it is possible to realize the arrangement of the virtual sound source desired by the viewer.

[0040] In one embodiment of the present invention, the signal processing device further includes a display unit for displaying a distance from the sound source to the viewer.

With this configuration, the viewer can visually recognize the distance between the virtual sound source and the viewer.

The recording medium of the present invention includes an audio data area for storing an audio signal, an image data area for storing an image signal, and navigation data indicating the arrangement of the audio data area and the image data area. A recording medium including a navigation data area for storing and an auxiliary data area for storing auxiliary data,
The sound signal correction data is stored in at least one of the sound data area, the image data area, the navigation data area, and the auxiliary data area, and the sound signal correction data is a correction of the sound signal. The method includes at least one of a correction command specifying a method and a filter coefficient for correcting the audio signal.

With this configuration, it is possible to correct the sound signal in connection with the reproduction of the image signal or the sound signal stored in the recording medium.

[0044] In one embodiment of the present invention, the correction command includes at least one of the sound data area, the image data area, and the navigation data area.
And the filter coefficient is stored in the auxiliary data area.

With this configuration, it is possible to prevent the reproduction of the image signal, the audio signal, or the navigation data from being hindered by reproducing the filter coefficient requiring a relatively large capacity as compared with the correction command.

In one embodiment of the present invention, at least one image pack is stored in the image data area, and the image pack includes the image signal and the sound signal correction data.

With this configuration, it is possible to change the method of correcting the sound signal according to the change of the image signal.

[0048] In one embodiment of the present invention, at least one sound pack is stored in the sound data area, and the sound pack includes the sound signal and the sound signal correction data.

With this configuration, it is possible to change the method of correcting the sound signal according to the change of the sound signal.

In one embodiment of the present invention, at least one navigation pack is stored in the navigation data area, and the navigation pack includes the navigation data and the sound signal correction data.

With this configuration, it is possible to change the method of correcting the sound signal in accordance with the change in the image signal or the change in the sound signal that changes based on the navigation data.

[0052]

Embodiments of the present invention will be described below with reference to the drawings. Note that the embodiment shown here is merely an example, and the present invention is not necessarily intended to be limited to this embodiment. Also,
In the following description, a DVD will be described as an example of a recording medium on which an image signal and an audio signal are recorded, but the recording medium used in the present invention is not limited to the DVD. As the recording medium, any type of recording medium other than DVD (for example, an optical disk other than DVD, a hard disk in a computer, etc.) can be used. Further, in the following description, an example will be described in which an image signal, an audio signal, or audio signal correction data is input to a signal processing device by reproducing a recording medium on which the image signal, the audio signal, or the audio signal correction data is recorded. However, the present invention is not limited to this. For example, by receiving a broadcast or communication, an image signal, an audio signal, or audio signal correction data included in the broadcast may be input to the signal processing device.

1. Configuration of Signal Processing Apparatus 1a FIG. 1a shows a configuration of a signal processing apparatus 1a according to an embodiment of the present invention. The signal processing device 1a is connected to a reproducing device 2 that reproduces information recorded on the DVD disk 1.

On the DVD disk 1, an audio signal AS, an image signal VS, navigation data, auxiliary data, audio signal correction data, and the like are recorded. The sound signal correction data includes a correction command for specifying a correction method of the sound signal AS, and at least one correction command for correcting the sound signal AS.
And two filter coefficients. Alternatively, the sound signal correction data may include only a correction command, or may include only at least one filter coefficient.

The correction data and the filter coefficients included in the audio signal correction data are input to the signal processing device 1a by reproducing the information recorded on the DVD disk 1 using the reproducing device 2. The details of the format of the DVD disk 1 will be described later with reference to FIGS.

The signal processing device 1a includes a memory 4 for storing a plurality of filter coefficients for correcting the acoustic signal AS;
Receiving the correction command from outside the signal processing device 1a,
Using a filter coefficient selection unit 3 that selects at least one of a plurality of filter coefficients stored in the memory 4 based on the correction command, and at least one filter coefficient selected by the filter coefficient selection unit 3, A correcting unit 5 for correcting the acoustic signal AS.

The memory 4 is configured to be able to receive at least one filter coefficient for correcting the acoustic signal AS from outside the signal processing device 1a.
The at least one filter coefficient input to the memory 4 is added to the plurality of filter coefficients stored in the memory 4. Alternatively, at least one filter coefficient input to the memory 4 may be replaced with at least one of a plurality of filter coefficients stored in the memory 4.

The sound signal AS corrected by the correction unit 5
Is output to the headphones 6. The headphones 6 convert the corrected acoustic signal AS into sound and output the sound.
The image signal VS output from the playback device 2 is output to the image display device 7 (for example, a television). The image display device 7 displays an image based on the image signal VS. Here, reference numeral 8 indicates a viewer who views an image displayed on the image display device 7 with the headphones 6 mounted.

FIG. 1B shows another embodiment in which the signal processing device 1a according to the embodiment of the present invention is used. In FIG. 1B, the same components as those shown in FIG. 1A are denoted by the same reference numerals, and description thereof will be omitted. In the example shown in FIG. 1b, the signal processing device 1a is connected to a receiver 2b that receives a broadcast signal. The receiver 2b can be, for example, a set-top box.

The broadcast can be, for example, a digital television broadcast. Alternatively, the broadcast may be a streaming broadcast over any network, including the Internet. Of course, the image signal, the sound signal, or the sound signal correction data received through such a broadcast may be temporarily stored in a recording medium (not shown) such as a hard disk, and the stored data may be input to a signal processing device. Good.

FIG. 1C shows still another embodiment in which the signal processing device 1a according to the embodiment of the present invention is used. In FIG. 1C, the same components as those shown in FIG. 1A are denoted by the same reference numerals, and description thereof will be omitted. FIG. 1c
In the example shown in (1), the signal processing device 1a is connected to a communication device 2c that receives a communication signal. Communication equipment 2c
May be, for example, a mobile phone that receives a communication signal through a wireless communication path. The communication device 2c may be a modem that receives a communication signal through a wired communication path.
Such a wireless communication path or a wired communication path can be connected to the Internet. The signal processing device 1a temporarily stores the image signal or the audio signal or the audio signal correction data received as the communication signal in a recording medium (not shown) such as a hard disk, and inputs the stored data to the signal processing device. You may.

Hereinafter, a case where the signal processing apparatus 1a is connected to a reproducing apparatus 2 for reproducing information recorded on a DVD disk 1 as shown in FIG.
Each component included in the signal processing device 1a will be described. However, this description also applies to the case where the signal processing device 1a is used in the usage pattern shown in FIGS. 1B and 1C. For example, the logical format of the DVD disk 1 described later can be applied to the logical format of the broadcast signal shown in FIG. 1B and the communication signal shown in FIG. 1C.

2. Logical Format of DVD Disk 1 FIG. 2 shows an example of a logical format of the DVD disk 1.

In the example shown in FIG.
Is provided with a data information recording area 10 for recording the volume and file structure of the DVD disk 1 and a multimedia data area 11 for recording multimedia data including still image data. The sound signal correction data 12a is stored in an area other than the data information recording area 10 and the multimedia data area 11.

The multimedia data area 11 is a DVD
A navigation data area 13 for recording information relating to the entire disc 1 and menu information common to the entire DVD disc 1, a still picture data area 14 for recording data relating to a still picture, and recording data relating to sound. And an audio data area 15 for performing the operation. The detailed structure of the still image data area 14 will be described later with reference to FIG. The detailed structure of the sound data area 15 will be described later with reference to FIG.

The navigation data area 13 includes an acoustic navigation area 16 and a still image navigation area 1.
7 and an acoustic navigation assistance area 18.

The acoustic navigation area 16 stores acoustic navigation data 19 and acoustic signal correction data 12b.

The still image navigation area 17 contains the still image navigation data 20 and the sound signal correction data 12.
c is stored.

The acoustic navigation auxiliary area 18 stores audio navigation auxiliary data 21 and audio signal correction data 12d.

As described above, the sound signal correction data 12b-
Each of 12d is stored on the DVD disk 1 so as to accompany the navigation data.

FIG. 3 shows an example of the logical format of the still picture data area 14 shown in FIG.

The still picture data area 14 includes a still picture information area 22, a still picture object recording area 23, and a still picture information auxiliary area 24.

The still picture information area 22 stores still picture information data 25 and sound signal correction data 12e.

In the still image object recording area 23, 1
The above-described still image set 26 is stored. The still image set 26 includes one or more still image objects 27. The still image object 27 includes a still image information pack 28 and a still image pack 29. The still image pack 29 includes still image data 30 and audio signal correction data 12f.

The still image information auxiliary area 24 stores the still image information auxiliary data 31 and the sound signal correction data 12g.

As described above, the sound signal correction data 12e to
Each of the 12g is stored on the DVD disk 1 so as to accompany the data regarding the still image.

FIG. 4 shows the sound data area 1 shown in FIG.
5 shows an example of the logical format 5.

The sound data area 15 includes the sound information area 32
And an audio object recording area 33 and an audio information auxiliary area 34.

The sound information area 32 contains sound information data 3
5 and the sound signal correction data 12h are stored.

The sound object recording area 33 stores one or more sound objects 36. The sound object 36 includes one or more sound cells 37. The sound cell 37 includes one or more sound packs 38 and one or more auxiliary information packs 39. The sound pack 38 stores the sound data 4
0 and the sound signal correction data 12i. The auxiliary information pack 39 includes the auxiliary information data 41 and the sound signal correction data 1
2j.

Here, the sound object 36 corresponds to one or more music pieces. The acoustic cell 37 indicates a minimum unit of the acoustic signal AS that can be reproduced and output by the reproducing device 2.
The sound pack 38 indicates one frame of the sound signal AS obtained by dividing the sound signal AS into frames at predetermined time intervals. The auxiliary information pack 39 contains the audio signal A
This section shows parameters and control commands used when reproducing S.

In the audio information auxiliary area 34, the audio information auxiliary data 42 and the audio signal correction data 12k are stored.

As described above, the sound signal correction data 12h-
Each of the 12k is stored on the DVD disk 1 so as to accompany data relating to sound.

FIG. 5 shows another example of the logical format of the DVD disk 1.

In the example shown in FIG.
Is provided with a data information recording area 51 for recording the volume and file structure of the DVD disk 1 and a multimedia data area 52 for recording multimedia data including moving image data. The sound signal correction data 12a is stored in an area other than the data information recording area 51 and the multimedia data area 52.

The multimedia data area 52 includes a navigation data area 53 for storing navigation data, and one or more image / sound data areas 54 for recording image and sound data. The detailed structure of the image / sound data area 54 will be described later with reference to FIG.

The navigation data contains information about the entire DVD disc 1 and / or the DVD disc 1
Shows the menu information (the arrangement of the sound data area and the image data area) common to all of the items. The image signal and the video signal change based on the navigation data.

The navigation data area 53 stores an image
It includes an audio navigation area 55, an image / audio object navigation area 56, and an image / audio navigation auxiliary area 57.

In the image / sound navigation area 55,
Image / sound navigation data 58 and sound signal correction data 12m are stored.

The image / sound object navigation area 56 stores image / sound object navigation data 60 and sound signal correction data 12p.

The image / sound navigation auxiliary area 57 stores image / sound navigation auxiliary data 59 and sound signal correction data 12n.

As described above, the sound signal correction data 12m
Each of the 12p is stored on the DVD 1 so as to accompany the navigation data.

FIG. 6 shows an example of the logical format of the image / sound data area 54 shown in FIG.

The image / sound data area 54 includes a control data area 61 for recording common control data in the image / sound data area 54 and a common menu for recording a common menu in the image / sound data area 54. AV object set menu area 62 and AV object recording area 63
And a control data auxiliary area 64 for recording common control auxiliary data in the image / audio data area 54.

The AV object recording area 63 stores one or more AV objects 65. The AV object 65 includes one or more AV cells 66. The AV cell 66 includes one or more AV object units 67. The AV object unit 67 is a navigation pack 6
8, A pack 69, V pack 70 and SP pack 71
By time division multiplexing.

The navigation pack 68 includes navigation data 72 having a pack structure and sound signal correction data 12q. A
The pack 69 includes sound data 73 having a pack structure and sound signal correction data 12r. The V pack 70 includes image data 74 having a pack structure and sound signal correction data 12s. The SP pack 71 includes sub-image data 75 having a pack structure and sound signal correction data 12t.

Here, the AV object 65 indicates an image signal VS and a sound signal AS for one track. The track is one unit of the image signal VS and the sound signal AS when the reproduction device 2 reproduces. The AV cell 66 indicates the minimum unit of the image signal VS and the sound signal AS that can be reproduced and output by the reproducing device 2.

As described above, the sound signal correction data 12q ~
Each of 12 t is stored on the DVD disc 1 so as to accompany data relating to image and sound.

Acoustic signal correction data 12a (FIGS. 2 and 5)
Is a DV in which the image signal VS and the sound signal AS are stored.
It is stored in an area different from the area on the D disk 1.
Therefore, it is possible to output the audio signal correction data 12a from the reproducing device 2 before outputting the image signal VS and the audio signal AS from the reproducing device 2. For example, a plurality of acoustic signal correction data 12a for correcting the acoustic characteristics of a plurality of types of headphones that can be used by the viewer 8 are stored in the DVD disk 1 in advance, and are stored in the headphones 6 actually used by the viewer 8. Corresponding sound signal correction data 12a
Is selected, and the audio signal AS is corrected using the selected audio signal correction data 12a, so that the audio signal A adapted to the headphones 6 actually used by the viewer 8 is selected.
The correction of S can be realized. Similarly, the audio signal correction data 12a for realizing the audio characteristics desired by the viewer 8 may be stored in the DVD disk 1 in advance.

The sound signal correction data 12c (FIG. 2)
And the sound signal correction data 12e to 12g (FIG. 3)
The data is stored on the DVD disk 1 so as to accompany the data related to the still image. Therefore, the sound signal correction data 12
c, 12e to 12g can be read from the DVD disk 1 at the timing when data relating to a still image is read. As a result, in synchronization with the output of the image signal VS from the playback device 2, the sound signal correction data 12c and 12e
To 12 g can be output from the playback device 2.
Thus, the sound signal AS can be corrected in conjunction with the content of the still image displayed on the image display device 7.

For example, when a recording location (for example, a concert hall, a studio, or the like) of the audio signal AS is displayed on the image display device 7, the audio signal correction data 12c for reproducing the sound field at the recording location, The sound signal AS may be corrected using 12e to 12g. As a result, it is possible for the viewer 8 to enjoy the acoustic characteristics suitable for the visual sense.

For example, when a near view or a distant view of a musical instrument or a singer is displayed on the image display device 7, the sound signal correction data 12c and 12e to reproduce the distance from the sound source to the viewer 8 are displayed.
The sound signal AS may be corrected using 12g. As a result, the viewer 8 can enjoy audio characteristics suitable for the visual sense.

Note that the audio signal AS is a DVD disc 1
By the creator (content creator) of the D so as to have an acoustic characteristic (acoustic characteristic suitable for the sight) corresponding to the still image.
It may be recorded on the VD disc 1. Such recording is generally performed by a content creator in a mixing room or a sound studio while adjusting sound characteristics. In this case, the sound signal AS may be corrected using the sound signal correction data 12c and 12e to 12g for reproducing the sound field of the mixing room or the sound studio. Thereby, the viewer 8 can enjoy the acoustic characteristics (acoustic characteristics suitable for the sight) adjusted by the content creator.

The sound signal correction data 12b, 12d
(FIG. 2) and the sound signal correction data 12h to 12k (FIG. 4) are stored on the DVD disc 1 so as to accompany the data related to sound. Therefore, it is possible to output the audio signal correction data 12b, 12d, 12h to 12k from the reproducing device 2 in synchronization with the output of the audio signal AS from the reproducing device 2. Thereby, the acoustic signal AS
The sound signal AS can be corrected in conjunction with the content of the sound signal AS.

For example, the sound signal AS may be corrected according to a song or a phrase of the song. As a result, the viewer 8 can enjoy suitable acoustic characteristics.

Further, the sound signal correction data 12m to 12m
t (FIGS. 5 and 6) is stored on the DVD disk 1 so as to accompany data including images and sounds including moving images. Therefore, the audio signal correction data 12m to 12t can be output from the reproduction device 2 in synchronization with the output of the image signal VS and the audio signal AS from the reproduction device 2. This makes it possible to correct the sound signal AS in conjunction with the content of the image (moving image) displayed on the image display device 7 and / or the content of the sound signal AS. As a result, the viewer 8 can enjoy the acoustic characteristics suitable for the visual sense.

3. Correction Command and Filter Coefficient FIG. 7 shows an example of a correction command and a filter coefficient included in audio signal correction data (for example, the audio signal correction data 12a shown in FIG. 2).

As shown in FIG. 7, the correction command 81
Is represented by, for example, two bits. In this case, the filter command stored in the memory 4 can be designated in four ways using the correction command 81. One filter coefficient may be designated by the correction command 81, or a plurality of filter coefficients may be designated.

The filter coefficient 82 is, for example, as shown in FIG.
Is one of the filter coefficients 83a to 83n or one of the filter coefficients 84a to 84n.

Each of the filter coefficients 83a to 83n indicates an "impulse response" (a transfer function indicating the acoustic characteristics of a direct sound) expressing the acoustic transfer characteristics from a sound source in a predetermined sound field to a listening point. Each of the filter coefficients 84a to 84n indicates a "reflected sound structure" (acoustic characteristics of reflected sound) expressing the level and time of a sound reaching a listening point in a predetermined sound field.

Note that each of the plurality of filter coefficients stored in the memory 4 also includes the filter coefficients 83a to 83n.
Or any of the filter coefficients 84a to 84n. Preferably, the memory 4 stores a plurality of different types of filter coefficients. Viewer 8
To provide acoustic characteristics of various sound fields.

For example, by using the filter coefficient 83a as the filter coefficient 82, the convolution operation of the impulse response corresponding to the filter coefficient 83a can be performed in the correction unit 5. As a result, the viewer 8 can listen to the sound reproducing the sound transfer characteristic from the sound source in the predetermined sound field to the listening point. The filter coefficient 82
By using the filter coefficient 84a as, the viewer 8 can receive a sound reproducing the reflected sound structure of a predetermined sound field.

When the correction command 81 is represented by two bits, the capacity required for recording the correction command 81 can be small. Therefore, the correction command 81
Recording on the disc 1 does not squeeze the capacity of the DVD disc 1.

The correction command 81 is stored in at least one of the navigation data area 13 (FIG. 2), the still picture data area 14 (FIG. 2), and the sound data area 15 (FIG. 2). The filter coefficient 82 is preferably stored in an area (for example, an auxiliary data area) other than the navigation data area 13, the still image data area 14, and the audio data area 15. As a result, by reproducing the filter coefficient 82 requiring a relatively large capacity as compared with the correction command 81, the image signal V
It is possible to prevent the reproduction of the S or the audio signal AS or the navigation data from being disturbed.

As described above, according to the signal processing device 1a of the present invention, the viewer 8 can hear the sound suitable for the image displayed on the image display device 7 from the headphones 6. The correction performed on the audio signal AS by the correction unit 5 changes following a change in the image signal VS and / or a change in the audio signal AS. As a result, the viewer 8 does not feel a sense of discomfort in the relationship between vision and hearing.

The filter coefficients in the memory 4 used for correcting the sound signal AS can be appropriately added, selected, and changed. Therefore, not only the sound signal AS is corrected in accordance with the change of the image signal VS and / or the change of the sound signal AS, but also the headphones 6 actually used by the viewer 8.
It is possible to correct the acoustic characteristics due to the acoustic characteristics of the viewer 8 and the physical individual differences of the viewer 8 (for example, the shape of the ears and face of the viewer 8).

Further, the capacity required to record the correction command 81 used to select the filter coefficient in the memory 4 is relatively small. As a result, by recording the correction command 81 on the DVD disk 1, the capacity of the DVD disk 1 is not squeezed. Also,
The time required to read the correction command 81 from the DVD disk 1 is also reduced. Therefore, it is possible to finely switch the filter coefficient according to a change in the image signal VS or a change in the sound signal AS. As a result, it is possible to change the method of correcting the audio signal AS in a manner that more reflects the intention of the creator of the image signal VS and the audio signal AS (content).

When receiving the broadcast signal or the communication signal and inputting the sound signal correction data included in the broadcast signal or the communication signal to the signal processing device, the correction is performed by receiving the broadcast signal or the communication signal. The command 81 is input to the signal processing device 1a. The bandwidth required to broadcast or communicate the correction command 81 is
Relatively small. As a result, the broadcast or communication of the correction command 81 does not reduce the bandwidth of the broadcast or communication.

As described above, by correcting the audio signal AS based on the audio signal correction data recorded on the DVD disk 1, the viewer 8 can obtain a favorable viewing environment.

In the present embodiment, the case where still image data and audio data are recorded on a DVD disk has been described. However, the same applies to the case where only audio data is recorded on a DVD disk. It is possible to correct the signal AS. In this case, it goes without saying that the same effect as described above can be obtained.

Further, in the present embodiment, the case where the filter coefficients included in the acoustic signal correction data recorded on the DVD disk 1 are stored in the memory 4 has been described, but the filter coefficients are stored in the memory 4 in advance. You may keep it. Alternatively, filter coefficients stored in a flexible disk or a semiconductor memory may be transferred to the memory 4. The filter coefficients may be input from the DVD disk 1 to the memory 4 as needed. In these cases, it is needless to say that the same effects as those described above can be obtained.

Further, in the present embodiment, the case where the correction command has two bits has been described, but the correction command is not limited to two bits. The number of bits of the correction command can be increased or decreased according to the type of the filter coefficient stored in the memory 4 and the capacity of the DVD disc 1. Further, the content of the correction command may be any content as long as a filter coefficient used for correcting the audio signal AS can be specified. In this case, it is needless to say that the same effects as those described above can be obtained.

Further, in the present embodiment, the case where a filter coefficient indicating an impulse response and a filter coefficient indicating a reflected sound structure are used as the filter coefficients has been described. Other types of filter coefficients may be used. In this case, it is needless to say that the same effects as those described above can be obtained. Further, it goes without saying that the impulse response and the reflected sound structure may be used together.

Furthermore, in the present embodiment, the case where the corrected sound signal AS is output to the headphones 6 has been described, but the output destination of the corrected sound signal AS is not limited to the headphones 6. The corrected sound signal AS is
It can be output to any type of converter (for example, a speaker) having a function of converting the electric acoustic signal AS into a wave. In this case, it is needless to say that the same effects as those described above can be obtained.

[0125] 4. Use of the buffer memory 87 without interrupting the output from the playback device 2
Is corrected, the signal processing device 1a (FIG. 1a)
It is preferable to have the buffer memory 87. Hereinafter, the use of the buffer memory 87 will be described.

FIGS. 8a to 8c and FIGS. 9a to 9c
FIG. 2 shows a state in which the image signal VS, the audio signal AS, and the audio signal correction data recorded on the DVD disk 1 are being reproduced by the reproducing device 2.

FIGS. 8A and 9A show the DVD disc 1
8b and FIG. 9b show the state temporally later than FIGS. 8a and 9a, respectively, and FIGS. 8c and 9c show the initial state immediately after the start of the reproduction of FIG. Shows a later state in time.

8a to 8c, reference numeral 85
Indicates an initial data area to be reproduced first after the reproduction of the DVD disk 1 is started, reference numeral 86 indicates a data area following the initial data area 85, and reference numeral 88
Indicates an area in which the acoustic signal correction data 12 is recorded.

Further, reference numeral 87 indicates a buffer memory for temporarily storing data reproduced from the initial data area 85 and sequentially outputting the stored data. Here, the buffer memory 87 is controlled so that the speed at which data is input to the buffer memory 87 is faster than the speed at which data is output from the buffer memory 87. For example, the speed at which data is output from the buffer memory 87 is a speed required for normal reproduction (1 × speed reproduction) of the image signal VS or the audio signal AS from the DVD disk 1, and the speed at which data is input to the buffer memory 87. Reproduces the image signal VS or the audio signal AS from the DVD disk 1 normally (1
Faster than required to play at double speed).

At least one filter coefficient included in the sound signal correction data recorded on the DVD disk 1 is stored in the memory 4 while the image signal VS or the sound signal AS is being output from the buffer memory 87.

The time required for outputting the image signal VS or the audio signal AS from the buffer memory 87 is equal to or longer than the time required for storing at least one filter coefficient included in the audio signal correction data in the memory 4. It is.

In the initial state shown in FIG. 8A, the data recorded in the initial data area 85 is reproduced at a higher speed than that required for normal reproduction. As a result, the image signal VS
The sound signal AS is input to the buffer memory 87 at a speed higher than the speed required for normal reproduction. Buffer memory 8
7 stores the output from the initial data area 85 and outputs the image signal VS and the sound signal AS stored in the buffer memory 87 at a speed required for normal reproduction.

When the output of data from the initial data area 85 is completed, the state transits from the initial state shown in FIG. 8A to the state shown in FIG. 8B.

In the state shown in FIG. 8B, the sound signal correction data 12 recorded in the area 88 is reproduced at a speed higher than that required for normal reproduction. As a result, the filter coefficients included in the acoustic signal correction data 12 are output to the memory 4. On the other hand, the buffer memory 87 stores the image signals VS stored in the buffer memory 87 at a speed required for normal reproduction.
And an audio signal AS.

When the output of the sound signal correction data 12 from the area 88 is completed, the state changes from the state shown in FIG. 8B to the state shown in FIG. 8C.

In the state shown in FIG. 8C, data recorded in the data area 86 following the initial data area 85 is reproduced at a speed required for normal reproduction. As a result, the correction command recorded in the data area 86 is output to the filter coefficient selection unit 3. The filter coefficient selection unit 3 outputs to the memory 4 a signal specifying a filter coefficient to be selected from among a plurality of filter coefficients stored in the memory 4 according to the correction command. The memory 4 outputs the filter coefficient specified by the filter coefficient selection unit 3 to the correction unit 5. The correction unit 5 corrects the audio signal AS using the filter coefficient output from the memory 4.

In the initial state shown in FIG. 9A, the data recorded in the initial data area 85 is reproduced at a higher speed than that required for normal reproduction. As a result, the image signal VS
The sound signal AS is input to the buffer memory 87 at a speed higher than the speed required for normal reproduction. Buffer memory 8
7 stores the output from the initial data area 85 and outputs the image signal VS and the sound signal AS stored in the buffer memory 87 at a speed required for normal reproduction.

Further, the correction command recorded in the initial data area 85 is output to the filter coefficient selecting section 3 via the buffer memory 87. The filter coefficient selection unit 3 outputs to the memory 4 a signal specifying a filter coefficient to be selected from among a plurality of filter coefficients stored in the memory 4 according to the correction command. The memory 4 outputs the filter coefficient specified by the filter coefficient selection unit 3 to the correction unit 5. The correction unit 5 corrects the audio signal AS using the filter coefficient output from the memory 4 (however, the processing from when the correction command is output to the filter coefficient selection unit 3 until when the audio signal AS is corrected is: FIG.
a is not shown). Since the filter coefficients are not reproduced from the initial data area 85, the initial data area 8
The acoustic signal AS recorded in 5 is output without being corrected (or is corrected using one of a plurality of filter coefficients stored in the memory 4 in advance).

When the output of the data from the initial data area 85 is completed, the state transits from the initial state shown in FIG. 9A to the state shown in FIG. 9B.

In the state shown in FIG. 9B, the sound signal correction data 12 recorded in the area 88 is reproduced at a higher speed than that required for normal reproduction. As a result, the filter coefficients included in the acoustic signal correction data 12 are output to the memory 4. On the other hand, the buffer memory 87 stores the image signals VS stored in the buffer memory 87 at a speed required for normal reproduction.
And a sound signal AS, and outputs a correction command to the filter coefficient selection unit 3.

When the output of the sound signal correction data 12 from the area 88 is completed, the state changes from the state shown in FIG. 9B to the state shown in FIG. 9C.

In the state shown in FIG. 9C, the data recorded in the data area 86 following the initial data area 85 is reproduced at a speed required for normal reproduction. As a result, the correction command recorded in the data area 86 is output to the filter coefficient selection unit 3. The filter coefficient selection unit 3 outputs to the memory 4 a signal specifying a filter coefficient to be selected from among a plurality of filter coefficients stored in the memory 4 according to the correction command. The memory 4 outputs the filter coefficient specified by the filter coefficient selection unit 3 to the correction unit 5. The correction unit 5 corrects the audio signal AS using the filter coefficient output from the memory 4.

As described above, by effectively using the buffer memory 87, the sound signal AS based on the sound signal correction data 12 can be corrected without interrupting the output of the image signal VS and the sound signal AS from the reproducing apparatus 2. It is possible to do.

In the present embodiment, the sound signal AS recorded in the initial data area 85 is output without correction (or one of a plurality of filter coefficients stored in the memory 4 in advance). Is corrected using one). Therefore, it is preferable that the initial data area 85 store the acoustic signal AS that does not need to be corrected. The initial data area 85 can store, for example, the title of the content (for example, a movie) of the DVD disc 1 and the audio signal AS and the image signal VS relating to an advertisement or the like provided by the content creator.

In the present embodiment, the case where the initial data area 85 is reproduced first after the reproduction of the DVD disk 1 is started has been described, but the audio signal correction data is reproduced after the reproduction of the DVD disk 1 is started. Even if the area 88 in which 12 is recorded is reproduced first,
The audio signal AS can be corrected based on the audio signal correction data 12 without interrupting the output of the image signal VS and the audio signal AS from the playback device 2.

In the present embodiment, the case where image data and audio data are recorded in the initial data area 85 has been described as an example. However, either one of the image data and the audio data, or other data is used. Data (for example, navigation data) is stored in the initial data area 85.
It is needless to say that the same effect as described above can be obtained also when the information is recorded in

[0147] 5. Configuration of Correction Unit 5 FIG. 10A shows an example of the configuration of the correction unit 5 (FIG. 1A).
The correction unit 5 shown in FIG. 10A controls the sound signal A according to at least one filter coefficient output from the memory 4.
Transfer function correction circuit 9 for correcting the transfer function of S
Including 1.

In the following description, it is assumed that sound waves in space are transmitted as shown in FIG.

In FIG. 11, reference numeral 94 indicates a space forming a sound field, reference numeral 95 indicates a sound source installed at a predetermined position, and C1 indicates a direct sound from the sound source 95 to the right ear of the viewer 8. C2 indicates a transfer characteristic of a direct sound from the sound source 95 to the left ear of the viewer 8, and R1 indicates a transfer characteristic of the sound source 9.
5 shows the transmission characteristics of the reflected sound from 5 to the right ear of the viewer 8;
R2 indicates the transfer characteristic of the reflected sound from the sound source 95 to the left ear of the viewer 8.

Hereinafter, with reference to FIG. 12, how the filter coefficient of the transfer function correction circuit 91 is determined when the viewer 8 listens to the sound through the headphones 6 will be described.

FIG. 12 shows an example of the configuration of the transfer function correction circuit 91.

Transfer function correction circuit 91 includes an FIR filter 96a and an FIR filter 96b. The acoustic signal AS is input to the FIR filters 96a and 96b.
The output of the FIR filter 96a is input to the right channel speaker 6a of the headphones 6. FIR filter 9
The output of 6 b is input to the left channel speaker 6 b of the headphones 6.

A case where the sound from the virtual sound source 95 is reproduced by the headphones 6 will be considered. Here, the transfer function of the FIR filter 96a is W1, and the transfer function of the FIR filter 96b is W1.
2. The transfer function from the right-channel speaker 6a of the headphone 6 to the right ear of the viewer 8 is Hrr, and the transfer function from the left-channel speaker 6b of the headphone 6 to the left ear of the viewer 8 is Hll. In this case, (Equation 1) holds.

[0154]

(C1 + R1) = W1 · Hrr (C2 + R2) = W2 · Hll The sound from the virtual sound source 95 is output to the headphone by using W1 and W2 obtained from (Equation 1) as transfer functions of the FIR filters 96a and 96b, respectively. 6 can be reproduced. That is, it is possible for the viewer 8 to actually feel the sound radiated from the headphones 6 as if the sound were radiated from the sound source 95.

From equation (1), the transfer function W1 of the FIR filter 96a and the transfer function W2 of the FIR filter 96b are
It is given by (Equation 2).

[0156]

## EQU2 ## W1 = (C1 + R1) / Hrr W2 = (C2 + R2) / Hll Hereinafter, referring to FIG.
How to determine the filter coefficient of the transfer function correction circuit 91 when listening to the sound via the 97b will be described.

FIG. 13 shows an example of the configuration of the transfer function correction circuit 91.

Transfer function correction circuit 91 includes an FIR filter 96a and an FIR filter 96b. The acoustic signal AS is input to the FIR filters 96a and 96b.
The output of the FIR filter 96a is input to the right channel speaker 97a, and is converted into sound by the speaker 97a. The output of the FIR filter 96b is input to the left channel speaker 97b, and is converted into sound by the speaker 97b.

The sound from the virtual sound source 95 is transmitted to the speaker 97a.
Consider the case of reproducing with 97b. Here, the transfer function of the FIR filter 96a is X1, the transfer function of the FIR filter 96b is X2, the transfer function from the speaker 97a to the right ear of the viewer 8 is Srr, and the transfer function from the speaker 97a to the left ear of the viewer 8. Is Srl, the transfer function from the speaker 97b to the right ear of the viewer 8 is Srl, and the transfer function from the speaker 97b to the left ear of the viewer 8 is Sll. In this case, (Equation 3) holds.

[0160]

(C1 + R1) = X1 · Srr + X2 · Slr (C2 + R2) = X1 · Srl + X2 · Sll X1 and X2 obtained from (Formula 3) are used as transfer functions of the FIR filters 96a and 96b, respectively, so that the virtual sound source 95 can be obtained. Can be reproduced by the speakers 97a and 97b. That is, actually, the speaker 97a,
It is possible to make the viewer 8 feel the sound radiated from the sound source 97b as if the sound were radiated from the sound source 95.

From equation (3), the transfer function X1 of the FIR filter 96a and the transfer function X2 of the FIR filter 96b are
It is given by (Equation 4).

[0162]

X1 = ｛Sll · (C1 + R1) −Slr ·
(C2 + R2)｝ / (Srr · Sll−Srl · Sl
r) X2 = ｛Srr · (C2 + R2) −Srl · (C1 + R
1)｝ / (Srr · Sll−Srl · Slr) FIG. 10B shows another example of the configuration of the correction unit 5 (FIG. 1A).

The correction unit 5 shown in FIG.
Transfer function correction circuit 91 for performing a process of correcting the transfer function of the audio signal AS according to at least one filter coefficient output from the memory 4, and at least one output from the memory 4
A reflected sound adding circuit 92 that performs a process of adding a reflected sound to the acoustic signal AS in accordance with the two filter coefficients, and an adder 93 that adds the output of the transfer function correction circuit 91 and the output of the reflected sound adding circuit 92.

The transfer function correction circuit 91 has a filter coefficient for reproducing the transfer characteristic of the direct sound from the sound source 95 to the viewer 8. The operation of the transfer function correction circuit 91 includes (C1 + R1) and (C2 + R) in (Equation 1) to (Equation 4).
2) except that each is replaced by C1 and C2,
It is the same as that shown in FIG. 10a. Therefore, the detailed description is omitted here.

The reflected sound adding circuit 92 has a filter coefficient for defining the level and time of the reflected sound reaching the viewer 8 after the sound emitted from the sound source 95 is reflected once or more.

FIG. 14 shows an example of the configuration of the reflected sound adding circuit 92.

As shown in FIG. 14, the reflected sound adding circuit 92 includes frequency characteristic adjusters 98a to 98n for adjusting the frequency characteristics of the acoustic signal AS, and frequency characteristic adjusters 98a to 98n.
Delay unit 99a for delaying the output of the terminal 98n by a predetermined time
To 99n and level adjusters 100a to 100n for performing predetermined gain adjustment on the outputs of the delay units 99a to 99n.
And an adder 101 for adding the outputs of the level adjusters 100a to 100n. The output of the adder 101 is the output of the reflected sound adding circuit 92.

In the frequency characteristic adjusters 98a to 98n, for example, the level of a certain frequency band component is raised or lowered,
By adding a low-pass filter or a high-pass filter characteristic, the frequency characteristic of the acoustic signal AS is adjusted.

In this way, the reflected sound adding circuit 92
The n (or n) independent reflected sounds are generated from the acoustic signal AS. By adjusting the frequency characteristic adjusters 98a to 98n, the delay units 99a to 99n, and the level adjusters 100a to 100n, it is possible to simulate the transfer characteristics R1 and R2 of the reflected sound in the space 94. This means that signals other than the direct sound can be realized by the reflected sound adding circuit 92.

The transfer function correction circuit 91 shown in FIG.
Can reduce the number of taps of the FIR filters 96a and 96b as compared with that shown in FIG. 10A.
This is because, unlike the case of FIG.
This is because 6a and 96b only need to express the transfer characteristics of the direct sound among the sounds reaching the viewer 8 from the sound source 95.

The operation time of the reflected sound adding circuit 92 is usually shorter than the operation time of the FIR filter having a large number of taps. Therefore, according to the configuration of FIG. 10B, the calculation time can be reduced as compared with the configuration of FIG. 10A.

Note that the frequency characteristic adjusters 98a to 98n,
Delay devices 99a to 99n, level adjusters 100a to 100
The order in which n is connected is not limited to the order shown in FIG. It is clear that the same effects as described above can be obtained even when these are connected in another order.

The number of frequency characteristic adjusters does not necessarily have to match the number of reflected sounds. For example, as shown in FIG. 15, the reflected sound adding circuit 92 is configured to include only the frequency characteristic adjuster 98a, and the frequency characteristic adjuster 9
8a: typical reflected sound characteristics (for example, frequency characteristics necessary to create a reflected sound with the highest gain)
May be corrected. Alternatively, as shown in FIG. 16, by setting average characteristics of similar reflected sounds, the number of frequency characteristic adjusters can be reduced.

Although not shown, a frequency characteristic adjuster is not used, and delay units 99a to 99n and a level adjuster 100 are used.
It is also possible to create a reflected sound only with a to 100n. In this case, although the accuracy of simulating the space 94 is reduced, an effect similar to the above-described effect can be obtained.

In FIGS. 15 and 16, the delay unit 9
It is apparent that the same effects as described above can be obtained even when 9a to 99n and the level adjusters 100a to 100n are connected in the reverse order.

FIG. 10C shows another example of the configuration of the correction unit 5 (FIG. 1A).

The correction unit 5 shown in FIG.
Transfer function correction circuit 91 for performing a process of correcting the transfer function of the audio signal AS in accordance with at least one filter coefficient output from the control unit, and at least one filter connected to the output of the transfer function correction circuit 91 and output from the memory 4 A reflection sound adding circuit 92 for performing a process of adding a reflection sound to the output of the transfer function correction circuit 91 in accordance with the coefficient.

The transfer function correction circuit 91 has a filter coefficient for reproducing the transfer characteristics of the direct sound from the sound source 95 to the viewer 8. The operation of the transfer function correction circuit 91 includes (C1 + R1) and (C2 + R) in (Equation 1) to (Equation 4).
2) except that each is replaced by C1 and C2,
It is the same as that shown in FIG. 10a. Therefore, the detailed description is omitted here.

The reflected sound adding circuit 92 has a filter coefficient for defining the level and time of the reflected sound reaching the viewer 8 after the sound emitted from the sound source 95 is reflected once or more.

FIG. 17 shows an example of the configuration of the reflected sound adding circuit 92.

The configuration of FIG. 17 is the same as the configuration of FIG. 14 except that the acoustic signal AS input to the reflected sound adding circuit 92 is input to the adder 101. Therefore, the same components are denoted by the same reference numerals, and description thereof will be omitted.

The sound signal AS is supplied to the frequency characteristic adjuster 98a.
９８98n and also to the adder 101. By making the output of the adder 101 the output of the correction unit 5, the sound from the virtual sound source 95 can be output to the headphone 6 or the speaker 97 in the same manner as in the case of FIGS.
a, 97b.

Further, frequency characteristic adjusters 98a to 98n
Since the input signal is an output signal of the transfer function correction circuit 91, a reflected sound in consideration of the transfer characteristic of the direct sound from the sound source 95 to the viewer 8 is added. This means
This is suitable for obtaining a sense of hearing as if the viewer 8 is the emission of sound from the sound source 95.

Note that the frequency characteristic adjusters 98a to 98n,
Delay devices 99a to 99n, level adjusters 100a to 100
The order in which n is connected is not limited to the order shown in FIG. It is clear that the same effects as described above can be obtained even when these are connected in another order.

The number of frequency characteristic adjusters does not necessarily have to match the number of reflected sounds. For example, as shown in FIG. 15, the reflected sound adding circuit 92 is configured to include only the frequency characteristic adjuster 98a, and the frequency characteristic adjuster 9
In step 8a, the characteristic of the representative reflected sound (for example, the correction characteristic of the frequency characteristic required to create the reflected sound with the largest gain) may be corrected. Alternatively, as shown in FIG. 16, by setting average characteristics of similar reflected sounds, the number of frequency characteristic adjusters can be reduced.

Although not shown, the frequency characteristic adjuster is not used, and the delay units 99a to 99n and the level adjuster 100 are used.
It is also possible to create a reflected sound only with a to 100n. In this case, although the accuracy of simulating the space 94 is reduced, an effect similar to the above-described effect can be obtained.

In FIGS. 15 and 16, the delay unit 9
It is apparent that the same effects as described above can be obtained even when 9a to 99n and the level adjusters 100a to 100n are connected in the reverse order.

In this embodiment, two reflected sounds R1 and R2 are shown. However, the same effect can be obtained when there are more reflected sounds.

Further, in the present embodiment, the case where there is one sound source 95 has been described, but when there are a plurality of sound sources 95, the above-described processing may be performed for each sound source. Thereby, the headphones 6 and the speakers 9 are actually
It is possible to make the viewer 8 feel the sound radiated from the sound sources 7a and 97b as if the sound were radiated from a plurality of sound sources.

6. Configuration of Filter Coefficient Selection Section 3 FIG. 18 shows an example of the configuration of the filter coefficient selection section 3 (FIG. 1A).

As shown in FIG. 18, a filter coefficient selecting section 3 automatically selects at least one of a plurality of filter coefficients stored in a memory 4 based on a correction command. 110 and a manual selection unit 111 for manually selecting at least one of the plurality of filter coefficients stored in the memory 4.

For example, as shown in FIG. 19, the manual selection section 111 includes a plurality of push-button switches 112a to 112a.
112n or the slide switch 1
13 or a rotary switch 11
4 may be provided. The viewer 8 can select at least one of the plurality of filter coefficients stored in the memory 4 by selecting a switch representing the desired signal processing. The selected filter coefficient is output to the correction unit 5.

Push button switches 112a to 112n
Means that the signal processing desired by the viewer 8 has discontinuous contents (for example, when performing sound processing for giving the acoustic characteristics of a hole to an audio signal, or when selecting a desired hole by the viewer 8). It is suitable for.

The slide switch 113 is used when the signal processing desired by the viewer 8 has continuous contents (for example,
When performing acoustic processing on the audio signal to give the viewer 8 a sense of hearing as if the direction and distance change as the sound source 95 moves, an auditory position of the sound source 95 desired by the viewer 8 is selected. It is suitable for

The rotary switch 114 is the same as the push-button switches 112a to 112n when the angle is changed every predetermined angle, and is the same as the slide switch 113 when the angle is changed continuously. Can be used.

By configuring the filter coefficient selection unit 3 as described above, the viewer 8 can not only obtain a hearing suitable for the sight based on the correction command, but also obtain a hearing desired by the viewer 8. become able to.

Note that the configuration of the filter coefficient selection unit 3 is not limited to the configuration shown in FIG. 18, and it is sufficient that the viewer 8 can select between signal processing desired by the viewer 8 and signal processing based on a correction command. For example, the configuration of the filter coefficient selection unit 3 may be the configuration shown in FIGS. 20A and 20B. In the configuration of FIGS. 20a and 20b, a function of determining which of the selection result by the manual selection unit 111 and the selection result by the automatic selection unit 110 has priority may be given to the manual selection unit 111. By selecting at least one of the plurality of filter coefficients stored in the memory 4 based on the determination result,
The same effect as the filter coefficient selection unit 3 shown in FIG.

7. FIG. 21 a is a top view showing the sound field 122 and FIG.
Is a side view showing the sound field 122. FIG.

As shown in FIG. 21A and FIG. 21B, a sound source 122 and a viewer 120 are arranged in a sound field 122. 21a and 21b, Pa is the sound source 12
1 shows a direct sound reaching directly to the viewer 120 from 1
Pb indicates the reflected sound reaching the viewer 120 after being reflected on the floor surface, Pc indicates the reflected sound reaching the viewer 120 after being reflected on the side surface, and Pn indicates the reflected sound reaching the viewer 120 after repeating the reflection a plurality of times. Shows the reflected sound reaching.

FIG. 22 shows reflected sound structures 123a to 123n obtained at the position of the left ear of the viewer 120 in the sound field 122 shown in FIG.

The sound radiated from the sound source 121
The direct sound Pa directly reaching the sound field 120 and the viewer 120 after being reflected by the wall surface (floor surface or side surface) surrounding the sound field 122.
, And reflected sounds Pb to Pn.

The sound radiated from the sound source 121 is
The time required to reach zero is proportional to the length of the path that the sound follows. Therefore, in the sound field 122 shown in FIGS. 21A and 21B, the direct sound Pa, the reflected sound Pb, the reflected sound Pc, and the reflected sound Pn reach the viewer 120 in this order.

[0203] The reflected sound structure 123a indicates the relationship between the arrival time of the direct sound Pa and the reflected sounds Pb to Pn reaching the position of the left ear of the viewer 120 from the sound source 121 and their levels. The vertical axis indicates level, and the horizontal axis indicates time. Time 0 on the horizontal axis represents the moment when sound is emitted from the sound source 121. Therefore, in the reflected sound structure 123a, the direct sound Pa is shown at the position closest to the time 0 in the order of reaching the position of the left ear of the viewer 120, and subsequently the reflected sound Pb, the reflected sound Pc, and the reflected sound Pn are shown. It is. As for the level of the reached sound, the direct sound Pa having the smallest distance attenuation and having no attenuation due to reflection is the largest.
The reflected sound is attenuated by distance and reflection, which increase according to the length of the path.
The level is shown decreasing as the process of reaching the left ear position of 0 is reached.

As described above, the reflected sound structure 123a has the sound field 1
Reference numeral 22 expresses the relationship between the time when the sound emitted from the sound source 121 reaches the position of the left ear of the viewer 120 and the level. Similarly, it is possible to obtain a reflected sound structure that expresses the relationship between the time at which the sound emitted from the sound source 121 in the sound field 122 reaches the position of the right ear of the viewer 120 and the level. The sound field 122 can be simulated by correcting the acoustic signal AS using the filter coefficients representing these reflected sound structures.

The reflected sound structures 123b to 123n correspond to the sound source 1
When the distance between the sound source 121 and the viewer 120 is gradually increased (the installation direction and the height of the sound source 121 viewed from the viewer 120 are not changed), the sound source 121 and the viewer 120 are not changed.
Direct sound Pa and reflected sound Pb reaching the left ear position
The relationship between the arrival time of Pn and its level is shown.

The distance between the sound source 121 and the viewer 120 in the case of the reflected sound structure 123b is longer than the distance between the sound source 121 and the viewer 120 in the case of the reflected sound structure 123a. The case of the sound structure 123b is behind the case of the reflected sound structure 123a. Similarly, the distance between the sound source 121 and the viewer 120 in the case of the reflected sound structure 123n is longer than the distance between the sound source 121 and the viewer 120 in the case of the reflected sound structure 123b. In the case of the sound structure 123n, the reflected sound structure 12
3b is behind.

[0207] The longer the distance between the sound source 120 and the viewer 121, the greater the distance attenuation. For this reason, the level of the reaching sound is lower in the case of the reflected sound structure 123b than in the case of the reflected sound structure 123a. Similarly, the level of the reaching sound is lower in the case of the reflected sound structure 123n than in the case of the reflected sound structure 123b.

Similarly, the arrival times of the reflected sounds Pb to Pn in the case of the reflected sound structures 123b to 123n are different from those of the reflected sound structure 12b.
3a, which is later than the arrival time of the reflected sounds Pb to Pn, and the reflected sound Pb in the case of the reflected sound structures 123b to 123n.
To Pn are lower than the levels of the reflected sounds Pb to Pn in the case of the reflected sound structure 123a. However, the level of the reflected sounds Pb to Pn of the reflected sound structures 123b and 123n is smaller than the level of the direct sound Pa. This is because the reflected sound has a longer path than the direct sound, and the ratio of the change in the path length due to the movement of the sound source 121 to the entire path length is smaller than that of the direct sound.

As in the case of the reflected sound structure 123a, the reflected sound structure 123b and the reflected sound structure 123n determine the time and level at which the sound radiated from the sound source 121 in the sound field 122 reaches the position of the left ear of the viewer 120. Expressing relationships.

[0210] Similarly, it is possible to obtain a reflected sound structure that expresses the relationship between the time and the level at which the sound radiated from the sound source 121 in the sound field 122 reaches the position of the right ear of the viewer 120. By correcting the acoustic signal AS using the filter coefficients expressing these reflected sound structures, the sound field 1
22 can be simulated.

Further, by selecting and using a plurality of reflected sound structures of the reflected sound structures 123a to 123n, the viewer 12
0 can hear the sound of the sound source at the desired position in the desired sound field.

In the above description, the case where there is one sound source has been described. Even when there are a plurality of sound sources, the sound field can be simulated by obtaining the reflected sound structure in the same manner. In the above description, the direction of arrival of the sound is not specified when obtaining the reflected sound structure.However, the simulation accuracy of the sound field can be further improved by specifying the direction of arrival of the sound and obtaining the reflected sound structure. Can be.

Hereinafter, a method for creating a reflected sound structure from the sound field 127 in which five speakers are arranged will be described with reference to FIG.

FIG. 23 shows a sound field 12 in which five sound sources are arranged.
FIG. 7 is a top view showing 7.

As shown in FIG. 23, sound sources 127a to 125e and a viewer 124 are arranged in a sound field 127. The sound sources 125a to 125e are arranged so as to keep the distance from the viewer 124 constant and surround the viewer 124. In FIG. 23, reference numeral 126
Reference numerals a to 126e denote ranges delimited by lines dividing the angle between adjacent sound sources into two with the viewer 124 as the center.

The sound sources 125a to 125e are arranged so as to form a general small-scale surround sound field. The sound source 125a is a sound source for a center channel installed in front of the viewer 124. The sound source 125b is the viewer 12
4 is a sound source for the right front channel installed diagonally forward right. The sound source 125c is a sound source for a front left channel installed diagonally forward left of the viewer 124. Sound source 125
d is a sound source for the rear right channel installed diagonally right behind the viewer 124. The sound source 125e is a sound source for the rear left channel installed diagonally left behind the viewer 124.

With the viewer 124 at the center, the sound source 125a
The angle between the sound source 125b and 125c is 30 degrees, and the angle between the sound source 125a and the sound source 125d or 125e is 120 degrees. The sound sources 125a to 125e are arranged in ranges 126a to 126e, respectively. The range 126a extends at 30 degrees around the viewer 124, the ranges 126b and 126c extend at 60 degrees around the viewer 124, and the ranges 126d and 12d.
6e is spread at 105 degrees around the viewer 124.

Hereinafter, the sound field 127 shown in FIG.
1a and the case where the sound field 122 shown in FIG. 21b is reproduced will be described as an example. The sound radiated from the sound source 121 in the sound field 122 follows various routes and reaches the viewer 120.
Therefore, the viewer 120 hears the direct sound coming from the direction of the sound source 121 and the reflected sounds coming from various directions. When such a sound field 122 is reproduced in the sound field 127, a reflected sound structure representing sound arriving at the positions of both ears of the viewer 120 in the sound field 122 is obtained for each direction of arrival, and used at the time of reproduction.

FIG. 24 shows the reflected sound structure 123a in the range 12
The reflected sound structure is shown for each direction of arrival of the sound divided into 6a to 126e. 24, reference numerals 128a to 128a
8e shows the reflected sound structure shown for the sound arriving from the ranges 126a to 126e, respectively.

FIG. 25 shows reflected sound structures 128a to 128e.
1 shows an example of the configuration of the correction unit 5 that reproduces the sound field 122 by using.

The correcting section 5 includes a transfer function correcting circuit 91 and reflected sound adding circuits 92a to 92e. The transfer function correction circuit 91 outputs the sound radiated from the sound source 125a to the viewer 12
4 is adjusted so that the sound characteristics when the sound radiated from the sound source 121 reaches the viewer 120 are equal to the sound characteristics when the sound reaches the viewer 120. Reflected sound adding circuits 92a to 92
“e” represents the reflected sound structures 128a to 128
Adjustment is made so as to create and output the same reflected sound as e.

By inputting the signals output from the reflected sound adding circuits 92a to 92e to the sound sources 125a to 125e, it is possible to simulate the sound field 122 with higher accuracy. This is because the reflected sound structures 128a to 128e can reproduce the level and time of the reflected sound in the sound field 122, and the sound sources 125a to 125e can reproduce the arrival direction of the reflected sound.

The same effect as described above can be obtained even when the transfer function correction circuit 91 is omitted from the configuration shown in FIG. It goes without saying that the transfer function correction circuit 91 does not necessarily have to be provided for the signal input to the sound source 125a.

In FIGS. 23 to 25, five sound sources 1
Although the case where the sound field 122 is reproduced with 25a to 125e has been described, five sound sources are not necessarily required. For example, it is also possible to reproduce the sound field 122 using headphones. This case will be described below.

FIG. 26 shows the sound field 12 using the headphones 6.
2 shows an example of the configuration of the correction unit 5 that reproduces No. 2.

As shown in FIG. 26, the correction section 5 includes a transfer function correction circuit 91 for correcting the acoustic characteristics of the acoustic signal AS.
a to 91j, reflected sound adding circuits 92a to 92j for adding reflected sounds to the outputs of the transfer function correcting circuits 91a to 91j,
An adder 129a for adding the outputs of the reflected sound adding circuits 92a to 92e and an adder 129b for adding the outputs of the reflected sound adding circuits 92f to 92j are included. The output of the adder 129a is input to the right channel speaker 6a of the headphones 6. The output of the adder 129b is input to the left channel speaker 6b of the headphones 6. 26, Wa to Wj are transfer function correction circuits 91a, respectively.
９１91j.

FIG. 27 shows a sound field 127 reproduced by the correction unit 5 shown in FIG. In the sound field 127, virtual sound sources 130a to 130e and a viewer 124 are arranged. The positions of the virtual sound sources 130a to 130e are shown in FIG.
Are the same as the positions of the sound sources 125a to 125e.

In FIG. 27, Cr indicates a transfer characteristic from the sound source 125a to the right ear of the viewer 124 when the viewer 124 does not wear the headphones 6, and Cl indicates that the viewer 124 wears the headphones 6. Sound source 12 when not present
5a shows the transfer characteristic from the viewer 124 to the left ear,
r indicates the transfer characteristic from the right channel speaker 6a of the headphone 6 to the right ear of the viewer 124, and Hl indicates the transmission characteristic from the left channel speaker 6b of the headphone 6 to the viewer 124.
5 shows the transfer characteristics up to the left ear of FIG.

A case where the sound from the sound source 125a is reproduced by the headphones 6 will be considered. Here, the transfer function correction circuit 91a
, The transfer function of the transfer function correction circuit 91f is Wf, the transfer function from the right channel speaker 6a of the headphone 6 to the right ear of the viewer 124 is Hr, and the left channel speaker 6b of the headphone 6 is the viewer. The transfer function to the left ear 124 is Hl. In this case, (Equation 5)
Holds.

[0230]

[Equation 5] Cr = Wa · Hr Cl = Wf · Hl The sound from the sound source 125a is reproduced by the headphones 6 by using Wa and Wf obtained from (Equation 5) as the transfer functions of the transfer function correction circuits 91a and 91f, respectively. It becomes possible to do. That is, it is possible to make the viewer 124 actually feel the sound radiated from the headphones 6 as if the sound were radiated from the sound source 125a.

From equation (5), the transfer function Wa of the transfer function correction circuit 91a and the transfer function Wf of the transfer function correction circuit 91f are obtained.
Is given by (Equation 6).

[0232]

## EQU6 ## Wa = Cr / Hr Wf = Cl / Hl The reflected sound adding circuit 92f extracts only the reflected sound arriving at the left ear of the viewer 124 from the direction of the range 126a represented by the sound source 125a. The reflected sound having 128a is added to the output signal of the transfer function correction circuit 91f. Similarly, the reflected sound adding circuit 92a transmits a reflected sound having a reflected sound structure (not shown) in which only the reflected sound arriving at the right ear of the viewer 124 is extracted from the direction of the range 126a represented by the sound source 125a. It is added to the output signal of the function correction circuit 91a. The reflected sound structure extracting only the reflected sound arriving at the right ear of the viewer 124 can be obtained by the same method as the method of obtaining the reflected sound structure 128a extracting only the reflected sound arriving at the left ear of the viewer 124. . As a result, the viewer 124
Not only obtains the auditory presence of the virtual sound source 130a,
A sound accurately simulating the direct sound and the reflected sound radiated from the sound source 125a can be received from the headphones 6.

Similarly, consider a case where the sound from the sound source 125b is reproduced by the headphones 6. Here, the transfer characteristic from the sound source 125b to the right ear of the viewer 124 when the viewer 124 does not wear the headphones 6 is Rr, and the viewer 12
4 is a sound source 125b when the headphones 6 are not worn
Is defined as Rl.
In this case, (Equation 7) holds.

[0234]

Rr = Wb · Hr Rl = Wg · Hl The sound from the sound source 125b is reproduced by the headphones 6 by using Wb and Wg obtained from (Expression 7) as transfer functions of the transfer function correction circuits 91b and 91g, respectively. It becomes possible to do. That is, it is possible for the viewer 124 to actually feel the sound radiated from the headphones 6 as if the sound were radiated from the sound source 125b.

From equation (7), the transfer function Wb of the transfer function correction circuit 91b and the transfer function Wg of the transfer function correction circuit 91g are obtained.
Is given by (Equation 8).

[0236]

Wb = Rr / Hr Wg = Rl / Hl The reflected sound adding circuit 92g extracts only the reflected sound arriving at the left ear of the viewer 124 from the direction of the range 126b represented by the sound source 125b. The reflected sound having 128b is added to the output signal of the transfer function correction circuit 91g. The reflected sound adding circuit 92b outputs a range 1 represented by the sound source 125b.
A reflected sound having a reflected sound structure (not shown) obtained by extracting only the reflected sound arriving at the right ear of the viewer 124 from the direction 26b is added to the output signal of the transfer function correction circuit 91b. The reflected sound structure extracting only the reflected sound arriving at the right ear of the viewer 124 can be obtained by the same method as the method of obtaining the reflected sound structure 128b extracting only the reflected sound arriving at the left ear of the viewer 124. . As a result, the viewer 124 not only obtains the auditory presence of the virtual sound source 130b, but also
A sound accurately simulating the direct sound and the reflected sound radiated from 5b can be received from the headphones 6.

In the same manner, the viewer 124 sets the transfer function correcting circuits 91c and 91h and the reflected sound adding circuits 92c and 9
2h allows the auditory presence of the virtual sound source 130c to be obtained, and the transfer function correction circuits 91d and 91i and the reflected sound adding circuits 92d and 92i allow the virtual sound source 130d to be obtained.
Can be obtained, and transfer function correcting circuits 91e and 91j and reflected sound adding circuits 92e and 92j can be obtained.
j makes it possible to obtain the auditory presence of the virtual sound source 130e.

As described above, by using the correction unit 5 shown in FIG. 26, the sound field 12 in which the sound sources 125a to 125e are arranged
7 can be reproduced. As a result, it is possible to reproduce the sound field 122 that can be reproduced in the sound field 127 using the correction unit 5.

[0239] In this embodiment, an example in which sound is received using headphones is described, but the present invention is not limited to this. For example, even when sound is received using two speakers, it is needless to say that the same effect as described above can be obtained by combining the transfer function correction circuit and the reflected sound adding circuit.

Further, in the present embodiment, an example in which the number of audio signals input to correction section 5 is one has been described, but the number of audio signals input to correction section 5 is limited to one. Do not mean. For example, the acoustic signal input to the correction unit 5 is
The sound signal may be a 5.1ch sound signal based on Dolby Surround.

Further, transfer function correction circuits 91a to 91j
Is connected to each of the reflected sound adding circuits 92a to 92j, and is not limited to the order shown in FIG. It is apparent that the same effect as described above can be obtained even when each of the transfer function correcting circuits 91a to 91j and each of the reflected sound adding circuits 92a to 92j are connected in the reverse order shown in FIG. .

FIG. 28 shows an example of the configuration of the correction unit 5 when the audio signal input to the correction unit 5 is a 5.1-channel audio signal based on Dolby Surround.

In the example shown in FIG. 28, a center channel signal (Center) radiated from a sound source installed in front of the viewer and a right channel signal radiated from a sound source installed diagonally forward right of the viewer. (Front
Right) and a left channel signal (Front Left) radiated from a sound source placed diagonally forward left of the viewer.
t) and a surround right channel signal (Surround) radiated from a sound source installed diagonally right behind the viewer.
Right) and a surround left channel signal (Sur) radiated from a sound source installed diagonally to the left of the viewer.
round Left) is input to the correction unit 5.

As shown in FIG. 28, each signal input to the correction unit 5 is corrected by using transfer function correction circuits 91a to 91j and reflection sound adding circuits 92a to 92j, so that the signals are actually output from the headphones 6. It is possible to make the viewer 124 feel the radiated sound as if it were the sound of each channel signal radiated from the virtual sound sources 130a to 130e.

Note that the reflected sound structure used in the reflected sound adding circuits 92a to 92j is not limited to the reflected sound structure obtained in the sound field 122. For example, by using a reflected sound structure obtained from a music hall desired by the viewer, the viewer 124 can be given a better hearing.

The sound signal input to the correction section 5 is
The above-mentioned center channel signal, right channel signal,
The present invention is not limited to a left channel signal, a surround right channel signal, and a surround left channel signal. For example, a signal such as a woofer channel signal or a surround back channel signal may be further input to the correction unit 5.
In this case, by correcting these signals using the transfer function correction circuit and the reflected sound adding circuit, the same effects as those described above can be obtained.

Further, in the present embodiment, a configuration has been described in which the acoustic signal input to the correction unit 5 is input to the transfer function correction circuit, and the output signal of the transfer function correction circuit is input to the reflection sound adding circuit. The acoustic signal input to the correction unit 5 may be input to the reflected sound adding circuit, and the output signal of the reflected sound adding circuit may be input to the transfer function correction circuit. In this case, the same effects as those described above can be obtained.

A range 1 that defines the direction of arrival of the reflected sound
26a to 126e are not limited to the above-described range, and may be changed depending on the sound field or the content of the acoustic signal.

For example, as shown in FIG.
Line L connecting the center position of the head of No. 24 and the center of the sound source 131
The range obtained by rotating the line Lb whose angle with a is θ degrees with respect to the line La in an axially symmetric manner (the range shown by the diagonal lines in FIG. 29) is used for obtaining the reflected sound structure used in the reflected sound adding circuit. The range defines the direction from which the reflected sound arrives. The reflected sound component included in the reflected sound structure increases as the angle θ between the line La and the line Lb increases, but the direction of arrival of the reflected sound obtained by the transfer function correction circuit and the reflected sound adding circuit is a simulated sound field. Therefore, the position of the virtual sound source becomes unclear. on the other hand,
The reflected sound component included in the reflected sound structure decreases as the angle θ between the line La and the line Lb decreases, but the direction of arrival of the reflected sound obtained by the transfer function correction circuit and the reflected sound adding circuit is simulated by a sound field. And the position of the virtual sound source becomes clear. The angle θ between the line La and the line Lb is 15
Degree is desirable. This is because the effect of the viewer's face and ear shape on the sound differs depending on the direction of arrival of the sound, and the characteristics of the sound to be heard differ.

FIG. 30 shows the results of measuring head-related transfer functions from the sound source to the subject's right ear in an anechoic room. In FIG. 30, HRTF1 indicates a head-related transfer function when one sound source is placed in front of the subject, and HRTF2 indicates 1
HRTF3 shows a head-related transfer function when one sound source is set at a position 30 degrees diagonally forward to the left of the subject. .

From FIG. 30, in the frequency band of 1 kHz or less, there is not much level difference in the measurement result.
It can be seen from FIG. In particular, HRT
There is a maximum level difference of about 10 dB between F1 and HRTF3. On the other hand, the level difference between HRTF1 and HRTF2 is at most about 3 dB.

FIG. 31 shows the results of measuring the head-related transfer function from the sound source to the subject's right ear for another subject different from the subject in FIG. Other measurement conditions such as the position of the sound source are the same between FIG. 30 and FIG. 31 except for the subject. In FIG. 31, HRTF4 indicates a head-related transfer function when one sound source is installed in front of the subject, and HRTF4 is shown.
Reference numeral 5 denotes a head-related transfer function when one sound source is set at a position 15 degrees to the left and to the front of the subject, and HRTF 6 denotes a head transfer when one sound source is set to a position 30 degrees to the left and at the front of the subject. Indicates a function.

HRTF1 (FIG. 30) and HRTF4 (FIG. 3)
HRTF2 (FIG. 30) and HRTF5
(FIG. 31) and HRTF3 (FIG. 30) and HRT
By comparing with F6 (FIG. 31), in the frequency band below the frequency (about 8 kHz) where a deep dip occurs, there is not much difference between the measurement result of FIG. 30 and the measurement result of FIG. 31, and the frequency band higher than 8 kHz It can be seen that the difference between the measurement result of FIG. 30 and the measurement result of FIG. 31 is large. This indicates that the characteristics of the subject greatly affect the head-related transfer function in a frequency band higher than 8 kHz. On the other hand, in the frequency band of 8 kHz or less, even if the subject changes, the head-related transfer function has the same measurement result if the sound source direction is the same. That is, when the transfer function correcting circuit and the reflected sound adding circuit simulate the sound field in which the direction of arrival is also considered for an unspecified number of viewers, 8 kHz
The characteristics of the simulated sound field may be reproduced for the following frequency bands. In the frequency band of 8 kHz or less, no significant difference is observed in the head-related transfer functions even when the sound source directions differ by 15 degrees.

In FIG. 29, the angle θ between the line La and the line Lb is 1
When the angle is set to 5 degrees or less, the viewer 124
The transfer function correction circuit is adjusted so as to have the transfer function up to and the reflected sound addition circuit is adjusted so as to have the reflected sound structure of the reflected sound incident from the shaded area in FIG. The reflected sound structure including more reflected sounds can be obtained while the position is clear. As a result, the accuracy of simulating the sound field is improved.

In the present embodiment, each of the ranges 126a to 126e defining the direction of arrival of the reflected sound has an angle formed by a line La connecting the center position of the head of the viewer 124 and the center of the sound source 131. It is assumed that the range is a range obtained by rotating the line Lb of θ degrees axially symmetrically with respect to the line La (a range indicated by oblique lines in FIG. 29). However, in each of the ranges 126a to 126e, as shown in FIG. 32A, a line La is set in the front direction from the ear position, and the line Lb having an angle of θ degrees with the line La is rotated axisymmetrically with respect to the line La. 32 (see FIG. 32)
32a, or a line La connecting the ear position and the center of the sound source 131 by setting a line La connecting the ear position and the center of the sound source 131 as shown in FIG. It may be a range obtained by being rotated axisymmetrically with respect to La (the range shown by oblique lines in FIG. 32B).

In the present embodiment, a method has been described in which a plurality of reflected sound structures (for example, reflected sound structures 123a to 123n) are selectively used to obtain a sense of distance to a sound source desired by a viewer. However, the reflected sound structure does not always need to be faithfully obtained from a simulated sound field. For example, as shown in FIG. 33, by expanding the time axis of the reflected sound structure 132a that obtains the closest sense of distance, the reflected sound structure 132k or the reflected sound structure 132n that obtains a sense of far distance.
And the reflected sound structure 13 for obtaining a sense of the farthest distance.
The reflected sound structure 133k and the reflected sound structure 133n that obtain a close sense of distance may be obtained by dividing and deleting 3a by a certain time width.

FIG. 34 shows another example of the configuration of the correction unit 5 when the audio signal input to the correction unit 5 is a 5.1-channel Dolby surround sound signal. 34, components that are the same as the components shown in FIG. 28 are given the same reference numerals, and descriptions thereof will be omitted. FIG.
In the example shown in FIG. 5, the correction unit 5 includes adders 143a to 143a.
3e. The adders 143a to 143e are used to input the output of the reflected sound adding circuit 92a to the transfer function correction circuits 91a to 91e. Transfer function correction circuits 91a-
The output of 91e is added by the adder 129a.
The output of the adder 129a is input to the right channel speaker of the headphones 6. According to the configuration of the correction unit 5 shown in FIG.
The reflected sound of the Center signal arriving at is accurately simulated.

Although FIG. 34 shows only components for generating a signal to be input to the right-channel speaker of the headphone 6, signals input to the left-channel speaker of the headphone 6 are the same. Needless to say, it can be generated as follows. Also, FIG.
Although the example of the configuration for accurately simulating the reflected sound of the er signal has been described, other signals (for example, Front Rig)
ht signal, FrontLeft signal, Surround
The correction unit 5 can also be configured to accurately simulate the reflected sound of the Right signal or the Surround Left signal).

Further, according to the configuration of the correction unit 5 described in the present embodiment, the transfer function correction circuit and the reflection The signal processing performed by the sound adding circuit can be different. Therefore, as shown in FIG. 35, a plurality of virtual sound sources 130a to 130e can be arranged at desired positions.

[0260] 8. Display of distance between virtual sound source and viewer As described above, a virtual sound source is created by the signal processing of correction unit 5. By changing the filter coefficient used in the correction unit 5, the distance between the virtual sound source and the viewer can be controlled. Therefore, by monitoring the change of the filter coefficient used in the correction unit 5,
The distance between the virtual sound source and the viewer can be displayed to the viewer.

FIG. 36 shows an example of displaying the distance between the virtual sound source and the viewer.

The display section 141 has lamps LE1 to LE6.
Is provided. The display unit 141 illuminates a lamp corresponding to the distance between the virtual sound source and the viewer in conjunction with the change in the filter coefficient used in the correction unit 5, thereby indicating the distance between the virtual sound source and the viewer to the viewer. indicate.

[0263] The display unit 142 is provided with a monitor M. The display unit 142 displays the distance between the virtual sound source and the viewer to the viewer by displaying the distance between the virtual sound source and the viewer by a number in conjunction with the change in the filter coefficient used in the correction unit 5. .

By providing the display unit 141 and the display unit 142 in the signal processing device 1a (FIG. 1a), the viewer can recognize the distance between the virtual sound source and the viewer by hearing and also visually recognize the virtual sound source and the viewer. You can recognize the distance to.

[0265] In this embodiment, an example in which six lamps are provided in display portion 141 has been described, but the number of lamps is not limited to six. As long as the viewer can recognize the distance between the virtual sound source and the viewer, it is obvious that any mode can be adopted as the display mode of the distance.

[0266]

According to the signal processing apparatus of the present invention, it is possible to change the method of correcting an audio signal according to a change in an image signal or a change in an audio signal. Thereby, the viewer can hear sound suitable for the image displayed on the image display device from the speaker or the headphones. As a result, it is possible to prevent the viewer from feeling uncomfortable with the relationship between the sight and the hearing.

Further, according to the signal processing device of the present invention, the sound processing is performed in accordance with the acoustic characteristics of the speakers or headphones used by the viewer and the acoustic characteristics based on the individual differences such as the shapes of the ears and faces of the viewer. It is possible to change the signal correction method. As a result, a better sound environment can be provided to the viewer.

According to the signal processing device of the present invention, the reproduction of the image signal, the audio signal, or the navigation data is hindered by reproducing the filter coefficient requiring a relatively large capacity as compared with the correction command. Is prevented.

According to the signal processing device of the present invention, it is possible to reproduce the sound signal correction data recorded on the recording medium without interrupting the image signal and the sound signal output from the reproducing device. .

Further, according to the signal processing device of the present invention, it is possible to make a viewer perceive a plurality of virtual sound sources using speakers or headphones, and to change the positions of the plurality of virtual sound sources. become. As a result, a sound field desired by the viewer can be created.

According to the signal processing device of the present invention, the distance between the virtual sound source and the viewer can be displayed to the viewer. This allows the viewer to recognize the distance between the virtual sound source and the viewer not only by hearing but also by vision.

[Brief description of the drawings]

FIG. 1a is a block diagram illustrating a configuration of a signal processing device 1a according to an embodiment of the present invention.

FIG. 1B is a block diagram showing another embodiment in which the signal processing device 1a according to the embodiment of the present invention is used;

FIG. 1c is a block diagram showing still another embodiment in which the signal processing device 1a according to the embodiment of the present invention is used.

FIG. 2 is a diagram showing an example of a logical format of a DVD disk 1;

FIG. 3 is a view showing an example of a logical format of a still image data area 14 shown in FIG. 2;

FIG. 4 is a view showing an example of a logical format of an acoustic data area 15 shown in FIG. 2;

FIG. 5 is a diagram showing another example of the logical format of the DVD disk 1;

FIG. 6 is a view showing an example of a logical format of an image / sound data area shown in FIG. 5;

FIG. 7 illustrates an example of a correction command and a filter coefficient.

FIG. 8A is a diagram showing a state where a signal recorded on a DVD disk 1 is being reproduced;

FIG. 8B is a diagram showing a state where a signal recorded on the DVD disk 1 is being reproduced.

FIG. 8C is a diagram showing a state where a signal recorded on the DVD disk 1 is being reproduced.

FIG. 9A is a diagram showing a state where a signal recorded on a DVD disk 1 is being reproduced.

FIG. 9B is a diagram showing a state where a signal recorded on the DVD disk 1 is being reproduced.

FIG. 9C is a diagram showing a state where a signal recorded on the DVD disk 1 is being reproduced.

FIG. 10A is a block diagram illustrating an example of a configuration of a correction unit 5;

FIG. 10B is a block diagram showing another example of the configuration of the correction unit 5;

FIG. 10C is a block diagram showing another example of the configuration of the correction unit 5;

11 is a top view showing the sound field 94. FIG.

FIG. 12 is a block diagram illustrating an example of a configuration of a transfer function correction circuit 91.

FIG. 13 is a block diagram illustrating an example of a configuration of a transfer function correction circuit 91.

FIG. 14 is a block diagram showing an example of the configuration of a reflected sound adding circuit 92.

FIG. 15 is a block diagram showing an example of the configuration of a reflected sound adding circuit 92.

FIG. 16 is a block diagram showing an example of the configuration of a reflected sound adding circuit 92.

FIG. 17 is a block diagram showing an example of the configuration of a reflected sound adding circuit 92.

FIG. 18 is a block diagram illustrating an example of a configuration of a filter coefficient selection unit 3.

FIG. 19 is a view for explaining the types of switches provided in the manual selection unit 111;

FIG. 20A is a block diagram showing an example of a configuration of a filter coefficient selection unit 3;

FIG. 20B is a block diagram illustrating an example of a configuration of a filter coefficient selection unit 3;

21a is a top view showing a sound field 122. FIG.

21b is a side view showing the sound field 122. FIG.

FIG. 22 is a diagram showing reflected sound structures 123a to 123n obtained at the position of the left ear of the viewer 120.

FIG. 23 is a top view showing a sound field 127 in which five sound sources are arranged.

FIG. 24 shows a reflection sound structure 123a in a range 126a to 126;
FIG. 4 is a diagram showing a reflected sound structure for each arrival direction of a sound sectioned by e

FIG. 25 is a block diagram showing an example of the configuration of a correction unit 5 that reproduces the sound field 122 using reflected sound structures 128a to 128e.

FIG. 26 is a block diagram showing an example of a configuration of a correction unit 5 that reproduces a sound field 122 using headphones 6.

FIG. 27 is a top view showing a sound field 127 reproduced by the correction unit 5 shown in FIG. 26;

FIG. 28 illustrates a case where the audio signal input to the correction unit 5 is a 5.1-channel audio signal based on Dolby Surround.
The block diagram which shows an example of a structure of the correction part 5.

FIG. 29 is a diagram illustrating an example of a range that defines an arrival direction of a reflected sound;

FIG. 30 is a view showing a result of measuring a head-related transfer function from a sound source to a subject's right ear in an anechoic room.

FIG. 31 is a view showing a result of measuring a head-related transfer function from a sound source to a subject's right ear for a subject different from the subject in FIG. 30;

FIG. 32A is a diagram illustrating an example of a range that defines an arrival direction of a reflected sound;

FIG. 32B is a diagram illustrating an example of a range that defines the direction of arrival of a reflected sound;

FIG. 33 is a diagram showing reflected sound structures 133a to 133n.

FIG. 34 shows a case where the audio signal input to the correction unit 5 is a 5.1-channel audio signal based on Dolby Surround.
FIG. 4 is a block diagram showing another example of the configuration of the correction unit 5

FIG. 35 is a diagram showing an arrangement of a plurality of virtual sound sources 130a to 130e.

FIG. 36 is a diagram illustrating an example of displaying a distance between a virtual sound source and a viewer.

[Explanation of symbols]

 REFERENCE SIGNS LIST 1 DVD disk 1 a signal processing device 2 playback device 3 filter coefficient selection unit 4 memory 5 correction unit 6 headphones 7 image display device 8 viewer

──────────────────────────────────────────────────続 き Continued on the front page (51) Int.Cl. ⁷ Identification symbol FI Theme coat ゛ (Reference) H04S 5/02 H04S 5/02 Y 7/00 F Z 7/00 G10K 15/00 B H04N 5/91 C (72) Inventor Hiroyuki Hashimoto 1006 Kadoma, Kadoma, Osaka Prefecture F-term in Matsushita Electric Industrial Co., Ltd.

Claims (23)

[Claims] 1. A signal processing device for processing an audio signal reproduced together with an image signal, wherein the memory stores a plurality of filter coefficients for correcting the audio signal, and specifies a method of correcting the audio signal. A filter coefficient selection unit that receives a correction command from outside the signal processing device and selects at least one of the plurality of filter coefficients stored in the memory based on the correction command; And a correction unit that corrects the acoustic signal using at least one filter coefficient selected by the method. 2. The correction command is input to the signal processing device by receiving a broadcast or a communication.
The signal processing device according to claim 1. 3. The signal processing device according to claim 1, wherein the correction command is recorded on a recording medium, and the correction command is input to the signal processing device by reproducing the recording medium. 4. The memory, wherein the memory receives at least one filter coefficient for correcting the acoustic signal from outside the signal processing device, and the received at least one filter coefficient is stored in the memory. And wherein the received at least one filter coefficient is configured to be replaced with at least one of the plurality of filter coefficients stored in the memory. Item 2. The signal processing device according to item 1. 5. The at least one filter coefficient,
The signal processing device according to claim 4, wherein the signal is recorded on a recording medium, and the at least one filter coefficient is input to the signal processing device by reproducing the recording medium. 6. The signal processing device further includes a buffer memory for temporarily storing the image signal and the audio signal, wherein a speed at which the image signal and the audio signal are input to the buffer memory is: The at least one filter coefficient recorded on the recording medium is higher than the speed at which the image signal and the audio signal are output from the buffer memory, and the image signal and the audio signal are output from the buffer memory. The time required for the image signal and the sound signal to be output from the buffer memory is required for the at least one filter coefficient to be stored in the memory. The signal processing device according to claim 5, wherein the time is equal to or longer than the time. 7. The at least one selected filter coefficient includes at least one filter coefficient representing a transfer function indicating an acoustic characteristic from a sound source to a viewer, and the correction unit represents the transfer function. The signal processing device according to claim 1, further comprising a transfer function correction circuit that corrects a transfer function of the acoustic signal according to the at least one filter coefficient. 8. The at least one selected filter coefficient includes at least one filter coefficient representing a transfer function indicating an acoustic characteristic of a direct sound from a sound source to a viewer, and a reflection from the sound source to the viewer. At least one filter coefficient representing a reflected sound structure indicating acoustic characteristics of sound, wherein the correction unit corrects a transfer function of the acoustic signal according to the at least one filter coefficient expressing the transfer function. A function correction circuit; a reflected sound adding circuit that adds a reflected sound to the acoustic signal according to the at least one filter coefficient representing the reflected sound structure; an output of the transfer function correcting circuit and an output of the reflected sound adding circuit The signal processing device according to claim 1, further comprising: an adder that adds 9. The at least one selected filter coefficient includes at least one filter coefficient representing a transfer function indicating an acoustic characteristic of a direct sound from a sound source to a viewer, and a reflection from the sound source to the viewer. At least one filter coefficient representing a reflected sound structure indicating acoustic characteristics of sound, wherein the correction unit corrects a transfer function of the acoustic signal according to the at least one filter coefficient expressing the transfer function. The signal processing device according to claim 1, further comprising: a function correction circuit; and a reflected sound adding circuit that adds a reflected sound to an output of the transfer function correction circuit according to the at least one filter coefficient representing the reflected sound structure. . 10. The filter coefficient selection unit, wherein the automatic selection unit automatically selects at least one of the plurality of filter coefficients stored in the memory based on the correction command; And a manual selection unit that manually selects at least one of the plurality of filter coefficients stored in the signal processing unit. 11. The at least one filter coefficient representing the reflected sound structure, wherein, when a distance between the sound source and the viewer is a first distance, a sound of a reflected sound from the sound source to the viewer. A first filter coefficient representing a reflected sound structure showing characteristics; and a reflected sound from the sound source to the viewer when a distance between the sound source and the viewer is a second distance different from the first distance. The signal processing device according to claim 8, further comprising: a second filter coefficient expressing a reflected sound structure indicating the acoustic characteristic of the signal. 12. The sound of a reflected sound from the sound source to the viewer when the distance between the sound source and the viewer is a first distance. A first filter coefficient representing a reflected sound structure showing characteristics; and a reflected sound from the sound source to the viewer when a distance between the sound source and the viewer is a second distance different from the first distance. The signal processing device according to claim 9, further comprising: a second filter coefficient expressing a reflected sound structure indicating the acoustic characteristic of the second filter. 13. The at least one filter coefficient representing the reflected sound structure is a third filter coefficient representing a reflected sound structure representing an acoustic characteristic of a reflected sound arriving at the viewer from a predetermined range of directions. The signal processing device according to claim 8, comprising: 14. The at least one filter coefficient representing the reflected sound structure is a third filter coefficient representing a reflected sound structure indicating an acoustic characteristic of a reflected sound arriving at the viewer from a direction in a predetermined range. The signal processing device according to claim 9, comprising: 15. The direction of the predetermined range is within a range of 15 degrees or less from a straight line connecting the sound source and the center of the viewer's head with the center of the viewer's head as the center. 14. The signal processing device according to claim 13, wherein 16. The direction of the predetermined range is within a range of 15 degrees or less from a straight line connecting the sound source and the center of the viewer's head with the center of the viewer's head as the center. The signal processing device according to claim 14, wherein 17. The sound signal includes sound signals of a plurality of channels, and the filter coefficient selection unit selects a filter coefficient corresponding to each of the sound signals of the plurality of channels.
The signal processing device according to claim 1. 18. The signal processing device according to claim 1, wherein the signal processing device further includes a display unit that displays a distance from the sound source to the viewer. 19. An audio data area for storing an audio signal, an image data area for storing an image signal, and navigation for storing navigation data indicating an arrangement of the audio data area and the image data area. A recording medium including a data area and an auxiliary data area for storing auxiliary data, wherein the audio signal correction data includes the audio data area, the image data area, the navigation data area, and the auxiliary data area. Stored in at least one of the audio signal correction data, the audio signal correction data includes at least one of a correction command specifying a correction method of the audio signal, and a filter coefficient for correcting the audio signal, recoding media. 20. The correction command is stored in at least one of the sound data area, the image data area, and the navigation data area, and the filter coefficient is stored in the auxiliary data area. The recording medium according to claim 19. 21. The image data area stores at least one image pack, wherein the image pack includes the image signal and the sound signal correction data.
The recording medium according to claim 19. 22. At least one sound pack is stored in the sound data area, and the sound pack includes the sound signal and the sound signal correction data.
The recording medium according to claim 19. 23. The navigation data area includes:
20. At least one navigation pack is stored, wherein the navigation pack includes the navigation data and the sound signal correction data.
A recording medium according to claim 1.