Sound field correction device, sound field correction method, and non-volatile storage medium

Abstract

The present application relates to a sound field correction device, a sound field correction method, and a non-volatile storage medium that effectively suppress sound deviation between a plurality of speakers. A sound field correction device is provided with: a test sound output unit that outputs a first test sound from a first speaker and a second speaker, outputs a second test sound from the first speaker when a first standby time has elapsed after the first test sound is output, and outputs the second test sound from the second speaker when a second standby time has elapsed after the first test sound is output; a calculation unit that calculates, on the basis of the recorded composite sound collected by the microphone, a first time difference between the timing at which the first test sound output by the first speaker reaches the microphone and the timing at which the first test sound output by the second speaker reaches the microphone, and a second time difference between the timing at which the first test sound output by the second speaker reaches the microphone; the second time difference is a second time difference between a timing at which the second test sound output by the first speaker reaches the microphone and a timing at which the second test sound output by the second speaker reaches the microphone.

Description

Sound field correction device, sound field correction method and non-volatile storage medium

[0001] CROSS-REFERENCE TO RELATED APPLICATIONS

[0002] This application claims priority to Japanese patent application No. 2023-022135, filed with the Japan Patent Office on February 16, 2023, and entitled “Sound field correction device, sound field correction method, and program,” and claims priority to Japanese patent application No. 2023-100624, filed with the Japan Patent Office on June 20, 2023, and entitled “Audio device, audio control method, and program,” the entire contents of which are incorporated herein by reference. Technical Field

[0003] The present application relates to a sound field correction device, a sound field correction method and a non-volatile storage medium. Background Art

[0004] In audio systems that output sound from multiple speakers, sound dispersion can occur because the sound from each speaker arrives at the listening position at different times. To address this issue, a technology exists that uses a microphone to capture test tones output from the speakers, calculates the distance between the speakers and the microphone based on the microphone's output signal, and adjusts the sound output timing accordingly.

[0005] In addition, the following sound localization technology is used in the device that outputs sound, which separates the correlated components that are correlated between the channels and the uncorrelated components that are not correlated between the channels from the stereo signal of the two channels. These correlated components and uncorrelated components are used to enable the user to perceive the position of the virtual sound source.

[0006] Prior art literature

[0007] Patent Literature

[0008] Patent Document 1: Japanese Patent Application Laid-Open No. 2007-110357

[0009] Patent Document 2: Japanese Patent Application Laid-Open No. 2003-333698

[0010] Summary of the Invention

[0011] The aforementioned sound variations are not only caused by the distance differences between each speaker and the listening position, but are also influenced by various factors, such as communication delays caused by interference in wireless communications and the time it takes to process data within the audio device. Conventional technology may not be able to fully suppress the sound variations caused by these various factors.

[0012] The problem to be solved by the embodiments of the present application is to provide a sound field correction device, a sound field correction method, and a program that can effectively suppress sound deviations among a plurality of speakers.

[0013] An embodiment of the present application is a sound field correction device that performs processing to reduce deviations in timing when sounds outputted from a plurality of speakers reach a predetermined listening position, and comprises: a test sound output unit that outputs a first test sound of a first frequency from a first speaker and a second speaker, and outputs a second test sound of a second frequency different from the first frequency from the first speaker when a first standby time has elapsed after the output of the first test sound, and outputs a second test sound from the second speaker when a second standby time different from the first standby time has elapsed after the output of the first test sound; a recording unit that collects sound using a microphone disposed at a listening position and records a composite sound, the composite sound including the first test sound and the second test sound outputted from the first speaker and the second speaker, respectively; and a calculation unit. , which calculates a first time difference and a second time difference based on a result of a frequency analysis process, wherein the frequency analysis process is a process capable of detecting a component corresponding to a first frequency and a component corresponding to a second frequency based on a recorded composite sound, the first time difference being the time difference between a timing when a first test sound output by a first speaker reaches a microphone and a timing when the first test sound output by a second speaker reaches the microphone, and the second time difference being the time difference between a timing when a second test sound output by a first speaker reaches the microphone and a timing when the second test sound output by a second speaker reaches the microphone; and a correction unit, which corrects at least one of the output timing of the sound from the first speaker or the output timing of the sound from the second speaker based on the first time difference and the second time difference.

[0014] Furthermore, in display devices such as televisions, when attempting to achieve sound localization by outputting sound from multiple built-in speakers, existing sound localization technology causes multiple sound images that should be separated to overlap, resulting in a decrease in sound quality and a lack of immersion.

[0015] Another embodiment of the present application aims to provide an audio device, an audio control method, and a program that can produce high-quality sound and images.

[0016] Another embodiment of the present application is an audio device that performs a process of outputting sound from a plurality of speakers. The audio device includes a separation unit, a sound image localization adjustment unit, and an output control unit. The separation unit separates an independent component, an uncorrelated component, and a correlated component from a left input signal corresponding to a left channel and a right input signal corresponding to a right channel, respectively. The independent component has no correlation between the left and right channels and corresponds to an end sound whose sound image is localized at the left or right end of a specified area. The uncorrelated component has no correlation between the left and right channels and corresponds to a background sound whose sound image range is larger than that of the end sound. The correlated component has a correlation between the left and right channels and corresponds to a center sound whose sound image is localized at a position inside the localized positions of the sound images of the left and right end sounds. The sound image localization adjustment unit adjusts at least one of the correlated component and the independent component so that the left and right sound images of the end sound generated by the independent component are localized at the left and right ends of the specified area, and the sound image of the center sound generated by the correlated component is localized between the localized positions of the left and right sound images of the end sound. The output control unit causes a plurality of speakers to output the end sound, the background sound, and the center sound. BRIEF DESCRIPTION OF THE DRAWINGS

[0017] 1 is a block diagram showing an example of a hardware configuration of an audio device according to an embodiment;

[0018] 2 is a block diagram showing an example of a functional configuration of a sound field correction device according to an embodiment;

[0019] 3 is a diagram illustrating an example of features of a first test tone and a second test tone according to the embodiment;

[0020] 4 is a diagram showing an example of frequency analysis processing performed on an acoustic signal of a composite sound according to the embodiment;

[0021] 5 is a diagram showing an example of a detection result of a first frequency according to an embodiment;

[0022] 6 is a diagram showing an example of a detection result of a second frequency according to the embodiment;

[0023] 7 is a diagram illustrating an example of a method for calculating the first time difference when the first time difference according to the embodiment is smaller than the output time of the first test tone;

[0024] FIG8 is a diagram showing an example of a method for determining a priority relationship of delays according to an embodiment;

[0025] 9 is a flowchart showing an example of processing in the sound field correction device according to the embodiment;

[0026] 10 is a diagram showing an example of a hardware configuration of a display device according to the first embodiment;

[0027] 11 is a diagram showing an example of the structure of a plurality of speakers involved in the first embodiment;

[0028] 12 is a diagram showing an example of a functional configuration of a DSP according to the first embodiment;

[0029] 13 is a diagram showing an example of the structure of a separation unit according to the first embodiment;

[0030] 14 is a diagram showing an example of components included in an input signal according to the first embodiment;

[0031] 15 is a diagram showing an example of a method for separating independent components and uncorrelated components from the difference between an Lch input signal and an Lch target signal according to the first embodiment;

[0032] 16 is a diagram showing an example of a method for separating independent components and uncorrelated components from the difference between the Rch input signal and the Rch target signal according to the first embodiment;

[0033] 17 is a diagram showing an example of the relationship between the output position of each sound and the sound image in the first embodiment;

[0034] 18 is a diagram showing an example of sound image localization adjustment processing performed on related components according to the first embodiment;

[0035] 19 is a flowchart showing an example of processing by DSP according to the first embodiment;

[0036] FIG20 is a diagram showing an example of the configuration of a plurality of speakers of the display device 1 according to the second embodiment;

[0037] FIG21 is a diagram showing an example of a functional configuration of a DSP according to a second embodiment;

[0038] 22 is a diagram showing an example of the relationship between the output position of each sound and the sound image in the second embodiment;

[0039] 23 is a diagram showing an example of sound image localization adjustment processing performed on independent components and uncorrelated components according to the second embodiment;

[0040] FIG. 24 is a flowchart showing an example of processing by the DSP according to the second embodiment.

[0041] DESCRIPTION OF NUMERALS 1-1···Audio device, 5···Sound field correction device, 11-1···CPU, 12-1···Memory, 13-1···Storage, 14-1···User I / F, 15-1···Communication I / F, 20···Communication bus, 21-1···Sound decoder, 22-1···Sound input ADC, 23-1···DSP, 25A···First delay circuit, 25B···Second delay circuit, 31A···Internal speaker, 31B···External speaker, 41···Remote controller, 45···Microphone, 46···Wireless modulation circuit, 47···Transmitter, 101-1 ···Sound output unit, 102-1···Test sound output unit, 103-1···Receiving unit, 104-1···Recording unit, 105-1···Calculation unit, 106-1···Correction unit, AS1, AS2···Sound, R···Target frequency domain, St···Acoustic signal, tr11···First rising time, tf11···First falling time, tr12···Second rising time, tf12···Second falling time, tr21···Third rising time, t f21···third falling time, tr22···fourth rising time, tf22···fourth falling time, TS1···first test tone, TS2···second test tone, W···time window, Δt···output time, Δt1···first time difference, Δt2···second time difference, Δtw1···first standby time, Δtw2···second standby time, 1···display device, 21···DSP (audio device), 23ML···left main speaker (first left speaker) Speaker), 23MR···right main speaker (first right speaker), 23SL···left speaker (second left speaker), 23SR···right speaker (second right speaker), 101···separation unit, 102···sound image localization adjustment unit, 103···output control unit, 201, 201R···adaptive filter (filtering unit), 211L, 211R···adaptive algorithm (coefficient updating unit), C, L1, L2, LL, R1, R2, RR, UU…sound image. DETAILED DESCRIPTION

[0042] Hereinafter, exemplary embodiments of the present application will be disclosed.

[0043] Fig. 1 is a block diagram showing an example of the hardware configuration of an audio device 1-1 according to an embodiment. The audio device 1-1 is a device capable of outputting sound from a plurality of speakers, and may be, for example, a stereo system, a video player, a recorder, a television, or a home theater system.

[0044] The audio device 1-1 according to this embodiment includes a sound field correction device 5, an internal speaker 31A, an external connection I / F (Interface) 32, and a remote controller 41. The audio device 1-1 is configured to be connectable to an external speaker 31B via the external connection I / F 32. The internal speaker 31A is an example of a first speaker, and the external speaker 31B is an example of a second speaker.

[0045] The sound field correction device 5 performs the following sound field correction processing: it adjusts the output timing of the sound AS1 output from the internal speaker 31A and the output timing of the sound AS2 output from the external speaker 31B to avoid sound deviation. The sound field correction processing is performed to reduce the deviation between the timing when the sound AS1 output from the internal speaker 31A reaches the user's listening position and the timing when the sound AS2 output from the external speaker 31B reaches the same listening position. The listening position in this embodiment is the location of the remote control 41.

[0046] The sound field correction device 5 involved in this embodiment includes a CPU (Central Processing Unit) 11-1, a memory 12-1, a storage 13-1, a user I / F 14-1, a communication I / F 15-1, a sound decoder 21-1, a sound input ADC (Analog to Digital Converter) 22-1, a DSP (Digital Signal Processor) 23-1, a first delay circuit 25A and a second delay circuit 25B. These components are connected to each other in a communicative manner via a communication bus 20.

[0047] CPU11-1 performs prescribed calculation processing and control processing according to the program (including firmware, application software, etc.) stored in the memory 12-1, etc. The memory 12-1 includes main storage devices such as RAM (Random Access Memory) and ROM (Read Only Memory), which function as a storage area for programs, a work area for CPU11-1, etc. The memory 13-1 is an auxiliary storage device including non-volatile memories such as SSD (Solid State Drive) and HDD (Hard Disk Drive), which can write and read various data. The user I / F 14-1 is a device that can receive user input and output information to the user, such as a display, input buttons, etc. The communication I / F 15-1 is a device that can realize communication with other electronic devices connected through a prescribed communication network. The communication I / F 15-1 of this embodiment establishes wireless communication with the remote control 41 according to the prescribed standard.

[0048] The audio decoder 21-1 converts audio data stored on a specified medium (e.g., CD, DVD, Blu-ray (registered trademark) disc, removable storage medium, etc.), audio data included in broadcast waves, or audio data obtained from a network such as a CSP (Communications Service Provider) into a digital signal that can be output from the internal speaker 31A and the external speaker 31B. The audio input ADC 22-1 converts analog audio signals input from an external device into digital signals.

[0049] DSP23-1 is a processor that performs prescribed processing on digital signals corresponding to the sounds output by the internal speaker 31A and the external speaker 31B to generate sound signals of the sounds AS1 and AS2 to be listened to, and sound signals of the first test sound TS1 and the second test sound TS2 to be described later.

[0050] The first delay circuit 25A delays the output timing of the sound output from the internal speaker 31A according to the correction signal (delay signal) output from the CPU 11-1. The second delay circuit 25B delays the output timing of the sound output from the external speaker 31B according to the correction signal output from the CPU 11-1.

[0051] The external connection I / F 32 transmits the audio signal generated by the DSP 23-1 and whose output timing is adjusted by the second delay circuit 25B to the external speaker 31B via a predetermined communication network. The communication network is not particularly limited and may be, for example, wireless communication conforming to a predetermined standard or wired communication using a predetermined cable. Examples of the predetermined standard include Bluetooth (registered trademark) and Wi-Fi (registered trademark).

[0052] The remote control 41 is a device that can be operated by users who want to listen to the sounds AS1 and AS2, and includes a microphone 45, a wireless modulation circuit 46, and a transmitter 47. The microphone 45 is a device that converts the collected sound into an electrical signal (analog signal). The wireless modulation circuit 46 is a circuit that modulates the electrical signal generated by the microphone 45 into a signal (digital signal) in a form that can be wirelessly communicated according to a prescribed standard. The transmitter 47 is a device that sends the signal modulated by the wireless modulation circuit 46 to the sound field correction device 5. It should be noted that, in addition to this, the remote control 41 is also provided with buttons for receiving user operations, etc., but the description is omitted here. In addition, in this embodiment, an example is given of a structure in which the remote control 41 is provided with a microphone 45, but the microphone 45 can be an independent device.

[0053] When performing sound field correction processing, the sound field correction device 5 of this embodiment causes the internal speaker 31A and the external speaker 31B to output a first test tone TS1 and a second test tone TS2, respectively. A microphone 45 mounted on the remote control 41 collects a composite sound containing the first test tone TS1 and the second test tone TS2. A transmitter 47 transmits an acoustic signal St of the composite sound collected by the microphone 45 to the sound field correction device 5. The sound field correction device 5 records the acoustic signal St received from the remote control 41 and performs frequency analysis on the recorded acoustic signal St. Based on the results of this frequency analysis, the sound field correction device 5 calculates (estimates) the difference in timing between the sounds AS1 and AS2 output from the internal speaker 31A and the external speaker 31B, respectively, when they arrive at the listening position. The device then corrects at least one of the output timings of the sound AS1 from the internal speaker 31A and the sound AS2 from the external speaker 31B to reduce this difference.

[0054] FIG2 is a block diagram illustrating an example of the functional configuration of a sound field correction device 5 according to an embodiment. The sound field correction device 5 according to this embodiment includes a sound output unit 101-1, a test sound output unit 102-1, a receiver 103-1, a recording unit 104-1, a calculation unit 105-1, and a correction unit 106-1. These functional units 101-1 through 106-1 can be implemented through the coordination of hardware and software (programs) as illustrated in FIG1 . Alternatively, at least a portion of these functional units 101-1 through 106-1 can be implemented using dedicated hardware (e.g., circuits).

[0055] The sound output unit 101 - 1 causes the internal speaker 31A and the external speaker 31B to output sounds AS1 and AS2 to be listened to, respectively.

[0056] When performing sound field correction processing, test sound output unit 102-1 causes internal speaker 31A and external speaker 31B to output a first test sound TS1 and a second test sound TS2, respectively. In this case, the frequency of first test sound TS1 and the frequency of second test sound TS2 differ from each other. Furthermore, the time interval (first standby time) between the first test sound TS1 and the second test sound TS2 output by internal speaker 31A differs from the time interval (second standby time) between the first test sound TS1 and the second test sound TS2 output by external speaker 31B.

[0057] Figure 3 illustrates an example of the characteristics of a first test tone TS1 and a second test tone TS2 according to an embodiment. As shown in Figure 3 , the first test tone TS1 is a sound at a first frequency Ft1 (e.g., approximately 5000 Hz), and the second test tone TS2 is a sound at a second frequency Ft2 (e.g., approximately 2000 Hz), which is different from the first frequency Ft1. While the output time Δt of the first test tone TS1 and the output time Δt of the second test tone TS2 are identical in this example, this is not a limitation. The specific value of this output time Δt is not particularly limited; for example, it can be approximately 50 ms.

[0058] Internal speaker 31A outputs second test tone TS2 after a first standby time Δtw1 has elapsed since the output of first test tone TS1. External speaker 31B outputs second test tone TS2 after a second standby time Δtw2, which is different from the first standby time Δtw1, has elapsed since the output of first test tone TS1. The specific values ​​of first standby time Δtw1 and second standby time Δtw2 are not particularly limited. For example, first standby time Δtw1 can be approximately 50 ms, and the difference between first standby time Δtw1 and second standby time Δtw2 can be approximately 20 ms.

[0059] For example, the output of the first test tone TS1 and the second test tone TS2 described above can be achieved by the following operation of test tone output unit 102-1. First, test tone output unit 102-1 simultaneously outputs a first instruction signal for outputting the first test tone TS1 to internal speaker 31A and external speaker 31B. Then, after a first waiting time Δtw1 has elapsed since the output of the first instruction signal, test tone output unit 102-1 outputs a second instruction signal for outputting the second test tone TS2 to internal speaker 31A. Furthermore, after a second waiting time Δtw2 has elapsed since the output of the first instruction signal, test tone output unit 102-1 outputs a third instruction signal for outputting the second test tone TS2 to external speaker 31B.

[0060] 2 , the receiving unit 103 - 1 receives the acoustic signal St of the composite sound collected by the microphone 45 placed at the listening position. The composite sound includes the first test sound TS1 and the second test sound TS2 described above. The receiving unit 103 - 1 of this embodiment receives the acoustic signal St transmitted from the remote controller 41 .

[0061] The recording unit 104-1 records the sound signal St received by the receiving unit 103-1 in a predetermined storage device (for example, the memory 13-1).

[0062] The calculation unit 105 performs frequency analysis on the acoustic signal St recorded by the recording unit 104-1. This allows it to detect components corresponding to the first frequency Ft1 (e.g., frequency components within a predetermined error range from 5000 Hz) and the second frequency Ft2 (e.g., frequency components within a predetermined error range from 2000 Hz) contained in the acoustic signal St. The specific method of frequency analysis is not particularly limited; however, an FFT (Fast Fourier Transform) can be utilized, for example. Based on the results of this frequency analysis, the calculation unit 105 calculates a first time difference Δt1 and a second time difference Δt2. The first time difference Δt1 is the time difference between the time when the first test tone TS1 output by the internal speaker 31A reaches the microphone 45 and the time when the first test tone TS1 output by the external speaker 31B reaches the microphone 45. The second time difference Δt2 is the time difference between the time when the second test tone TS2 output by the internal speaker 31A reaches the microphone 45 and the time when the second test tone TS2 output by the external speaker 31B reaches the microphone 45.

[0063] Correction unit 106-1 corrects at least one of the output timing of sound AS1 from internal speaker 31A or the output timing of sound AS2 from external speaker 31B based on the first time difference Δt1 and the second time difference Δt2 calculated by calculation unit 105. In this case, correction unit 106-1 of this embodiment determines the order of arrival of first test tone TS1 output by internal speaker 31A at microphone 45 and the arrival of first test tone TS1 output by external speaker 31B at microphone 45, based on the magnitude relationship between first time difference Δt1 and second time difference Δt2. In other words, the order of delay between internal speaker 31A and external speaker 31B. Based on this order of arrival, correction unit 106-1 relatively delays either the output timing of sound AS1 from internal speaker 31A or the output timing of sound AS2 from external speaker 31B to reduce first time difference Δt1.

[0064] The above configuration can effectively suppress sound deviation at the listening position due to various factors such as the distance difference between speakers 31A, 31B and the listening position, communication delay due to interference in wireless communication, and data processing time in the audio device.

[0065] FIG4 is a diagram illustrating an example of frequency analysis processing performed on an acoustic signal St of a composite sound according to an embodiment. In the graph illustrated in the upper portion of FIG4 , the horizontal axis (Time) represents the time elapsed from a predetermined reference time (e.g., when recording of the composite sound begins upon outputting an instruction signal (first instruction signal) for outputting the first test sound TS1), and the vertical axis (Intensity) represents the signal intensity of the composite sound collected by microphone 45. The graph illustrated in the lower portion of FIG4 shows the analysis result F of frequency analysis processing performed on the acoustic signal St. The horizontal axis (Frequency) corresponds to the frequency components contained in the composite sound, and the vertical axis (Power) corresponds to the power spectrum. Here, the analysis result F is illustrated when FFT processing is performed on a time window W corresponding to a predetermined time frame set in the acoustic signal St.

[0066] In this embodiment, the power value corresponding to the predetermined target frequency range R is calculated based on the analysis result F. Regarding the target frequency range R, when detecting (extracting) the first frequency Ft1, the target frequency range R is set based on the first frequency Ft1, and when detecting the second frequency Ft2, the target frequency range R is set based on the second frequency Ft2. For example, when detecting the first frequency Ft1, the target frequency range R is set to (Ft1±x%). For example, when Ft1=5000 (Hz) and x=5, R=5000±250 (Hz). Similarly, when extracting the second frequency Ft2, the target frequency range R is set to (Ft2±x%). For example, when Ft2=2000 (Hz) and x=20, R=2000±400 (Hz). By the above-described processing, the power value corresponding to the first frequency Ft1 (for example, R=5000±250) and the power value corresponding to the second frequency Ft2 (R=2000±400) can be obtained for each of the multiple time windows W that gradually deviate at a predetermined time interval.

[0067] FIG5 is a diagram showing an example of a detection result F1 of a first frequency Ft1 according to an embodiment. FIG6 is a diagram showing an example of a detection result F2 of a second frequency Ft2 according to an embodiment. In the detection result F1, the horizontal axis corresponds to time, and the vertical axis corresponds to the power value corresponding to the target frequency domain R (e.g., 5000±250) of the first frequency Ft1. In the detection result F2, the horizontal axis corresponds to time, and the vertical axis corresponds to the power value corresponding to the target frequency domain R (e.g., 2000±400) of the second frequency Ft2. The time axis of the detection result F1 is the same as the time axis of the detection result F2.

[0068] FIG5 illustrates a first rising moment tr11, a first falling moment tf11, a second rising moment tr12, and a second falling moment tf12 corresponding to the first frequency Ft1. The first rising moment tr11 is the first moment when the power value corresponding to the target frequency domain R of the first frequency Ft1 changes from a state less than the threshold value Th1 to a state greater than the threshold value Th1. The first falling moment tf11 is the first moment when the power value corresponding to the target frequency domain R of the first frequency Ft1 changes from a state greater than the threshold value Th1 to a state less than the threshold value Th1. The second rising moment tr12 is the second moment when the power value corresponding to the target frequency domain R of the first frequency Ft1 changes from a state less than the threshold value Th1 to a state greater than the threshold value Th1. The second falling moment tf12 is the second moment when the power value corresponding to the target frequency domain R of the first frequency Ft1 changes from a state greater than the threshold value Th1 to a state less than the threshold value Th1.

[0069] Here, let's assume that first test tone TS1 output by internal speaker 31A reaches microphone 45 before first test tone TS1 output by external speaker 31B. In this case, first rising time tr11 is the time when first test tone TS1 output by internal speaker 31A reaches microphone 45, and second rising time tr12 is the time when first test tone TS1 output by external speaker 31B reaches microphone 45. Therefore, the time from first rising time tr11 to second rising time tr12 is the first time difference Δt1.

[0070] FIG6 illustrates the third rising time tr21, third falling time tf21, fourth rising time tr22, and fourth falling time tf22 corresponding to the second frequency Ft2. The third rising time tr21 is the first time when the power value corresponding to the target frequency domain R of the second frequency Ft2 changes from less than the threshold value Th2 to more than the threshold value Th2. The third falling time tf21 is the first time when the power value corresponding to the target frequency domain R of the second frequency Ft2 changes from more than the threshold value Th2 to less than the threshold value Th2. The fourth rising time tr22 is the second time when the power value corresponding to the target frequency domain R of the second frequency Ft2 changes from less than the threshold value Th2 to more than the threshold value Th2. The fourth falling time tf22 is the second time when the power value corresponding to the target frequency domain R of the second frequency Ft2 changes from more than the threshold value Th2 to less than the threshold value Th2.

[0071] Here, it is assumed that second test tone TS2 output by internal speaker 31A reaches microphone 45 before second test tone TS2 output by external speaker 31B. In this case, third rising time tr21 is the time when second test tone TS2 output by internal speaker 31A reaches microphone 45, and fourth rising time tr22 is the time when second test tone TS2 output by external speaker 31B reaches microphone 45. Therefore, the time from third rising time tr21 to fourth rising time tr22 is the second time difference Δt2.

[0072] In the above description, as shown in Figure 5, a case is described where the first falling time tf11 precedes the second rising time tr12 on the time axis. In this case, the first time difference Δt1 can be calculated based on the first rising time tr11 and the second rising time tr12, as described above. However, if the first time difference Δt1 is less than the output time Δt of the first test tone TS1 (see Figure 3), the first wave W1 corresponding to the first test tone TS1 that arrives at microphone 45 first overlaps with the second wave W2 corresponding to the first test tone TS1 that arrives at microphone 45 later, and the first falling time tf11 and the second rising time tr12 do not occur. The following describes how to calculate the first time difference Δt1 in this case.

[0073] FIG7 illustrates an example method for calculating the first time difference Δt1 according to an embodiment when the first time difference Δt1 is less than the output time Δt of the first test tone TS1. In this case, as shown in FIG7 , since the first wave W1 and the second wave W2 overlap, only the first rising time tr11 and the second falling time tf12 appear on the time axis. In this case, the first time difference Δt1 can be calculated by subtracting the output time Δt of the first test tone TS1 from the time Δt' from the first rising time tr11 to the second falling time tf12.

[0074] The above calculation method also applies to the second time difference Δt2. That is, if the second time difference Δt2 is less than the output time Δt of the second test tone TS2, the second time difference Δt2 can be calculated by subtracting the output time Δt of the second test tone TS2 from the time from the third rising time tr21 to the fourth falling time tf22.

[0075] Furthermore, as described above, the correction unit 106 - 1 of the present embodiment determines the precedence relationship of the delay between the internal speaker 31A and the external speaker 31B based on the magnitude relationship between the first time difference Δt1 and the second time difference Δt2 .

[0076] FIG8 illustrates an example of a method for determining the order of delays according to an embodiment. FIG8 illustrates the magnitude relationship between the first time difference Δt1 and the second time difference Δt2, the order of delays, and the relationship between the speakers subject to delay control. The value a in FIG8 represents the difference between the first waiting time Δtw1 and the second waiting time Δtw2 (a = Δtw2 - Δtw1).

[0077] As shown in FIG8 , when first time difference Δt1 is greater than or equal to second time difference Δt2 (Δt1≧Δt2), it is determined that first test sound TS1 from external speaker 31B has reached microphone 45 before first test sound TS1 from internal speaker 31A. In this case, correction unit 106-1 delays the timing of outputting sound AS2 from external speaker 31B.

[0078] If first time difference Δt1 is smaller than second time difference Δt2 and first time difference Δt1 is less than a / 2 (Δt1 < Δt2 and Δt1 < a / 2), it is determined that first test sound TS1 from external speaker 31B has reached microphone 45 before first test sound TS1 from internal speaker 31A. In this case, correction unit 106-1 delays the timing of outputting sound AS2 from external speaker 31B.

[0079] When first time difference Δt1 is less than second time difference Δt2 and first time difference Δt1 is greater than or equal to a / 2 (Δt1 < Δt2 and Δt1 ≥ a / 2), correction unit 106-1 determines that first test sound TS1 from internal speaker 31A has reached microphone 45 before first test sound TS1 from external speaker 31B. In this case, correction unit 106-1 delays the timing of outputting sound AS1 from internal speaker 31A.

[0080] The above-mentioned determination method is merely an example, and the method for determining the order of delay is not limited to the above-mentioned method.

[0081] FIG9 is a flowchart illustrating an example of processing in the sound field correction device 5 according to the embodiment. When the sound field correction process begins, the test sound output unit 102-1 simultaneously outputs a first instruction signal for outputting the first test sound TS1 to the internal speaker 31A and the external speaker 31B ( S101-1 ). The test sound output unit 102-1 then determines whether a first waiting time Δtw1 has elapsed since the output of the first instruction signal ended ( S102-1 ). If the first waiting time Δtw1 has not elapsed ( S102-1 : No), the determination in step S102-1 continues. If the first waiting time Δtw1 has elapsed ( S102-1 : Yes), the test sound output unit 102-1 outputs a second instruction signal for outputting the second test sound TS2 to the internal speaker 31A ( S103-1 ). Next, test tone output unit 102-1 determines whether the second waiting time Δtw2 has elapsed since the output of the first instruction signal ended ( S104-1 ). If the second waiting time Δtw2 has not elapsed ( S104-1 : No), the determination in step S104-1 is continued. If the second waiting time Δtw2 has elapsed ( S104-1 : Yes), test tone output unit 102-1 outputs a second instruction signal to external speaker 31B for outputting second test tone TS2 ( S105-1 ). Through the above process, internal speaker 31A outputs first test tone TS1 and second test tone TS2 after the first waiting time Δtw1, and external speaker 31B outputs first test tone TS1 and second test tone TS2 after the second waiting time Δtw2.

[0082] A composite sound consisting of first test tone TS1 and second test tone TS2, respectively output by internal speaker 31A and external speaker 31B, is collected by microphone 45 at the listening position, and the collected composite sound is recorded (S106-1). Then, calculation unit 105 performs frequency analysis on the recorded composite sound to detect a first frequency component corresponding to first test tone TS1 and a second frequency component corresponding to second test tone TS2 (S107-1). Based on these detection results, calculation unit 105 calculates a first time difference Δt1 and a second time difference Δt2 (S108-1). As described above, first time difference Δt1 is the time difference between the time when first test tone TS1 output by internal speaker 31A arrives at microphone 45 and the time when first test tone TS1 output by external speaker 31B arrives at microphone 45. Second time difference Δt2 is the time difference between the time when second test tone TS2 output by internal speaker 31A arrives at microphone 45 and the time when second test tone TS2 output by external speaker 31B arrives at microphone 45.

[0083] Next, correction unit 106-1 determines the delay amount and the order of delay between internal speaker 31A and external speaker 31B based on the first time difference Δt1 and the second time difference Δt2 calculated as described above (S109-1). Based on this determination, correction unit 106-1 corrects at least one of the output timing of sound AS1 from internal speaker 31A or the output timing of sound AS2 from external speaker 31B to reduce the delay amount (S110-1).

[0084] According to the above embodiment, a composite sound including a first test tone TS1 and a second test tone TS2 outputted from two speakers (in this embodiment, internal speaker 31A and external speaker 31B), respectively, is recorded. Based on the analysis results of the recorded composite sound, the sound delay between the two speakers 31A and 31B is detected, and the output timing of sounds AS1 and AS2 from the two speakers 31A and 31B is corrected to reduce this delay. This effectively reduces sound deviations at the listening position caused by various factors.

[0085] Furthermore, according to this embodiment, the delay between two speakers 31A and 31B is detected using two test tones, including a first test tone TS1 and a second test tone TS2, with different wait times Δtw1 and Δtw2 between the first and second test tones TS1 and TS2. This method eliminates the need to output test tones of different frequencies from each speaker 31A and 31B, simplifying the structure for generating test tones.

[0086] Furthermore, while the above embodiment illustrates the case of correcting the sound deviation between internal speaker 31A and external speaker 31B, the number or type of speakers is not limited to this. For example, correction between two internal speakers or between two external speakers can also be performed using the same method as described above. Furthermore, if there are three or more speakers, the sound deviation between all speakers can be reduced by sequentially performing the same process on two of them.

[0087] The program for realizing the functions of the sound field correction device 5 described above can be provided in a computer-installable or computer-executable format, stored on a computer-readable storage medium such as a CD-ROM, floppy disk (FD), CD-R, or DVD (Digital Versatile Disk). Alternatively, the program can be stored on a computer connected to a network such as the Internet and provided by downloading the program via the network. Alternatively, the program can be provided or distributed via a network such as the Internet.

[0088] Another exemplary embodiment of the present application is disclosed below.

[0089] (First embodiment)

[0090] FIG10 is a diagram showing an example of the hardware configuration of a display device 1 according to the first embodiment. The display device 1 is a device capable of displaying images and outputting sound, and may be, for example, a television. The display device 1 illustrated here includes a processor 11, memory 12, a UI (User Interface) 13, peripheral circuits 14, a communication circuit 15, an audio decoder 16, an audio input ADC (Analog Digital Converter) 17, a DSP (Digital Signal Processor) 21 (an example of an audio device), an amplifier 22, and a speaker 23.

[0091] The processor 11 performs predetermined calculations and control operations according to programs stored in the memory 12. The memory 12 includes primary storage devices such as RAM (Random Access Memory) or ROM (Read Only Memory) and secondary storage devices such as SSD (Solid State Drive) or HDD (Hard Disk Drive), and stores programs and various data required to implement the functions of the display device 1. The UI 13 is a unit that receives user operations. The peripheral circuits 14 are circuits that assist the operations of the processor 11 and may include, for example, a power supply circuit, an oscillator circuit, a reset circuit, and the like.

[0092] The communication circuit 15 is a circuit that acquires video signals and audio signals through an antenna, a communication network, etc. The audio decoder 16 is a device that decrypts encrypted audio signals. The audio input ADC 17 is a device that converts analog audio signals into digital signals.

[0093] The DSP 21 is an information processing device that performs predetermined processing on audio signals received from the communication circuit 15, audio decoder 16, or audio input ADC 17. The DSP 21 may be comprised of, for example, a CPU (Central Processing Unit), memory, and input / output ports. The DSP 21 of this embodiment performs processing to optimize the audio output of each of the multiple speakers 23, so that the audio output from the multiple speakers 23 built into the display device 1 forms a predetermined sound image. The audio signals processed by the DSP 21 are output to each speaker 23 via the amplifier 22.

[0094] Fig. 11 is a diagram showing an example of the configuration of the plurality of speakers 23 according to the first embodiment. Fig. 11 illustrates a state where the display device 1 is viewed from the front side of the display 31.

[0095] As shown in FIG11 , the display device 1 of this embodiment includes a left main speaker 23ML (an example of a first left speaker), a right main speaker 23MR (an example of a first right speaker), a left speaker 23SL (an example of a second left speaker), and a right speaker 23SR (an example of a second right speaker). The left main speaker 23ML is located near the left end of the bottom of the display device 1 (display 31). The right main speaker 23MR is located near the right end of the bottom of the display device 1. The left speaker 23SL is located on the left side of the display device 1 and is positioned to the left of the left main speaker 23ML. The right speaker 23SR is located on the right side of the display device 1 and is positioned to the right of the right main speaker 23MR.

[0096] It should be noted that the arrangement of the left main speaker 23ML, the right main speaker 23MR, the left speaker 23SL, and the right speaker 23SR is an example and is not limited to the above configuration.

[0097] FIG12 is a diagram showing an example of the functional configuration of the DSP 21 according to the first embodiment. The DSP 21 of this embodiment includes a separation unit 101, a sound image localization adjustment unit 102, and an output control unit 103. These functional units are formed by the collaboration of the hardware and software that constitute the DSP 21. Furthermore, at least one of these functional units may be formed by dedicated hardware (e.g., circuits).

[0098] The separation unit 101 separates independent components, uncorrelated components, and correlated components from the Lch input signal and the Rch input signal, which are audio signals input to the DSP 21. The Lch input signal corresponds to the left channel in two-channel stereo playback, and the Rch input signal corresponds to the right channel in the stereo playback.

[0099] Independent components are components that have no correlation between the left and right channels and correspond to edge sounds, the sound image of which is localized at the left or right end of the display 31 (an example of a predetermined area). Uncorrelated components are components that have no correlation between the left and right channels and correspond to background sounds, the sound image of which is wider than that of edge sounds. Correlated components are components that have a correlation between the left and right channels and correspond to center sounds, the sound image of which is localized inward of the localized position of the sound image of the edge sounds.

[0100] The sound image localization adjustment unit 102 performs sound image localization adjustment processing to localize the sound image of the sound output from the multiple speakers 23 (in this embodiment, the left main speaker 23ML, the right main speaker 23MR, the left speaker 23SL, and the right main speaker 23MR) at an appropriate position. The sound image localization adjustment processing in this embodiment includes the following processing: adjusting at least one of the correlated components and the independent components so that the left and right sound images of the end sounds are localized at the left and right ends of the display 31 (display device 1), and the sound image of the middle sound is localized between the left and right localized positions of the end sounds.

[0101] The output control unit 103 causes the end sound generated by the independent components, the background sound generated by the uncorrelated components, and the middle sound generated by the middle components to be output from the corresponding speakers 23. In this embodiment, the left speaker 23SL outputs the Lch end sound generated by the independent components separated from the Lch input signal and the Lch background sound generated by the uncorrelated components separated from the Lch input signal. The right speaker 23SR outputs the Rch end sound generated by the independent components separated from the Rch input signal and the Rch background sound generated by the uncorrelated components separated from the Rch input signal. The left main speaker 23ML outputs the Lch middle sound generated by the correlated components separated from the Lch input signal. The right main speaker 23MR outputs the Rch middle sound generated by the correlated components separated from the Rch input signal.

[0102] 13 is a diagram showing an example of the configuration of the separation unit 101 according to the first embodiment. The separation unit 101 shown here includes adaptive filters 201L and 201R (an example of a filter unit) and adaptive algorithms 211L and 211R (an example of a coefficient update unit).

[0103] Adaptive filters 201L and 201R apply finite impulse responses (FIRs) based on predetermined coefficients xl0 to xl7 and xr0 to xr7 to the left and right input signals il and ir, respectively, to produce output signals ol and or. Specifically, the Lch adaptive filter 201L, corresponding to the left channel, applies a FIR based on coefficients xl0 to xl7 to the Lch input signal il, thereby outputting the output signal ol contained in the Lch input signal. The Rch adaptive filter 201R, corresponding to the right channel, applies a FIR based on coefficients xr0 to xr7 to the Rch input signal ir, thereby outputting the output signal or contained in the Rch input signal. It should be noted that while the example here illustrates a FIR with eight taps, this is not limiting.

[0104] Adaptive algorithms 211L and 211R are used to optimize the coefficients xl0 to xl7 and xr0 to xr7 of adaptive filters 201L and 201R. Adaptive algorithms 211L and 211R update the coefficients xl(n) and xr(n) to minimize the error signal values ​​el(n) and er(n). These error signal values ​​are based on the differences between the target signals tl and tr, which are the differences between the Lch input signal il and the Rch input signal ir, and the output signals ol and or from adaptive filters 201L and 201R. In this example, n is a value between 0 and 7, corresponding to the number of taps (8) of adaptive filters 201L and 201R. Lch adaptive algorithm 211L, corresponding to the left channel, updates the coefficients xl(n) to minimize the error signal value el(n). The error signal value el(n) is the difference between the value of the predetermined weight gl integrated with the target signal tl (the difference (il - ir) between the Lch input signal il and the Rch input signal ir) and the output signal ol output from the Lch adaptive filter 201L. The Rch adaptive algorithm 211R corresponding to the right channel updates the coefficient xr(n) to minimize the error signal value er(n). The error signal value er(n) is the difference between the value of the predetermined weight gr integrated with the target signal tr (the difference (ir - il) between the Rch input signal ir and the Lch input signal il) and the output signal or output from the Rch adaptive filter 201R.

[0105] Adaptive filters 201L and 201R, whose coefficients x10 to x17 and xr0 to xr7 are optimized by adaptive algorithms 211L and 211R as described above, separate independent components and uncorrelated components corresponding to error signals el(n) and er(n) from input signals il and ir. The independent and uncorrelated components are then subtracted from the input signals to separate the correlated components. Furthermore, the above-described structure eliminates the need for filtering (eliminating) uncorrelated components by adaptive filters 201L and 201R, reduces the risk of sudden changes in the filter's frequency characteristics, and suppresses the generation of components that could cause abnormal noise.

[0106] FIG14 is a diagram showing an example of components included in input signals il and ir according to the first embodiment. In this example, the Lch input signal il includes an uncorrelated component UL, an independent component IL, and correlated components CC, CL1, CL2, CR1, and CR2, and the Rch input signal ir includes an uncorrelated component UR, an independent component IR, and correlated components CC, CL1, CL2, CR1, and CR2.

[0107] The uncorrelated component UL included in the Lch input signal i1 is uncorrelated with the Rch input signal ir and corresponds to background sound. The uncorrelated component UR included in the Rch input signal ir is uncorrelated with the Lch input signal i1 and corresponds to background sound. For example, background sound is sound whose sound image is localized throughout the left and right areas of the display 31.

[0108] The independent component IL contained in the Lch input signal il is unrelated to the Rch input signal ir and is a component corresponding to the end sound on the left side. The independent component IR contained in the Rch input signal ir is unrelated to the Lch input signal il and is a component corresponding to the end sound on the right side. For example, the end sound is a sound whose sound image is localized at the left or right end of the display 31 or near the end. For example, the end sound on the left localizes the sound image at the left end of the display 31, and the end sound on the right localizes the sound image at the right end of the display 31. In Figure 14, the value accumulated with the independent components IL and IR, that is, the weight 1.0, represents the difference in output level (volume) of the sound between the left and right channels. That is, the end sound on the left side corresponding to the independent component IL is output entirely from the left channel side, not from the right channel side, and the end sound on the right side corresponding to the independent component IR is output entirely from the right channel side, not from the left channel side.

[0109] The associated components CC, CL1, CL2, CR1, and CR2 are components that are associated between the Rch input signal ir and the Lch input signal il and correspond to the center sound. For example, the center sound is a sound whose sound image is localized to the inside of the localization position of the left and right sound images of the end sound. The associated component CC is a component that corresponds to a sound whose sound image is localized at the center position within the left and right sound image range of the center sound. The associated component CL1 is a component that corresponds to a sound whose sound image is localized to the left of the center position within the sound image range. The associated component CL2 is a component that corresponds to a sound whose sound image is localized to the left of the localization position of the sound of the associated component CL1 within the sound image range and to the right of the localization position of the sound image to the left of the end sound. The associated component CR1 is a component that corresponds to a sound whose sound image is localized to the right of the center position within the sound image range. The correlation component CR2 corresponds to a sound whose sound image is localized to the right of the sound localization position of the correlation component CR1 and to the left of the sound image localization position on the right side of the end sound within the sound image range.

[0110] In FIG14 , the values ​​accumulated with the associated components CC, CL1, CL2, CR1, and CR2, i.e., the weights (0.90, 0.80, 0.70, 0.60, and 0.44 in this example), respectively, represent the output level difference of the sound between the left and right channels. That is, the output level of the left channel of the middle sound corresponding to the associated components CL1 and CL2 is greater than the output level of the right channel. In addition, the output level difference between the left and right channels in the associated component CL2 is greater than the output level difference between the left and right channels in the associated component CL1. The same is true for the associated components CR1 and CR2. It should be noted that if the weight of the left channel is set to Wl and the weight of the right channel is set to Wr, the weight setting illustrated here is Wl. 2 +Wr 2 becomes 1 or approximately 1. For example, the left and right weights of the correlation component CL1 become 0.90 2 +0.44 2 ≈1.0, the left and right weights of the associated component CL2 become 0.80 2 +0.64 2 = 1.0. The same also applies to the correlation components CR1 and CR2.

[0111] Based on the difference between the input signals il, ir and the target signals tl, tr as described above, the independent components IL, IR and the uncorrelated components UL, UR can be separated.

[0112] FIG15 is a diagram illustrating an example of a method for separating independent components IR and uncorrelated components UR by taking the difference between the Lch input signal i1 and the Lch destination signal t1 according to the first embodiment. The Lch destination signal t1 is obtained by subtracting the Rch input signal ir from the Lch input signal i1 as described above. As shown in FIG15 , since both the Lch input signal i1 and the Lch destination signal t1 include the Lch independent component IL, the Lch uncorrelated component UL, and the correlated components CL1, CL2, CR1, and CR2, the Rch independent component IR and the Rch uncorrelated component UR included in the Rch input signal ir can be separated by taking the difference between the Lch input signal i1 and the Lch destination signal t1.

[0113] FIG16 is a diagram illustrating an example of a method for separating an independent component IL and an uncorrelated component UL from the difference between an Rch input signal ir and an Rch destination signal tr according to the first embodiment. The Rch destination signal tr is the signal obtained by subtracting the Lch input signal il from the Rch input signal ir as described above. As shown in FIG16 , since both the Rch input signal ir and the Rch destination signal tr include an independent Rch component IR, an uncorrelated Rch component UR, and correlated components CL1, CL2, CR1, and CR2, the Lch independent component IL and the Lch uncorrelated component UL included in the Lch input signal il can be separated by taking the difference between the Rch input signal ir and the Rch destination signal tr.

[0114] By finding the difference between the Lch independent component IL and the Lch uncorrelated component UL separated as described above and the Lch input signal il, the Lch-correlated components CC, CL1, CL2, CR1, and CR2 corresponding to the left channel can be separated. Furthermore, by finding the difference between the Rch independent component IR and the Rch uncorrelated component UR separated as described above and the Rch input signal ir, the Rch-correlated components CC, CL1, CL2, CR1, and CR2 corresponding to the right channel can be separated.

[0115] FIG17 is a diagram illustrating an example of the relationship between the output position of each sound and its sound image in the first embodiment. In FIG17 , sound image LL represents the sound image on the left side of the end sound, and sound image RR represents the sound image on the right side of the end sound. Sound image UU represents the sound image of the background sound. Sound images C, L1, L2, R1, and R2 represent the sound image of the center sound.

[0116] The output control unit 103 of this embodiment causes the left speaker 23SL to output the Lch end sound generated based on the Lch independent component IL, and causes the right speaker 23SR to output the Rch end sound generated based on the Rch independent component IR. This allows the sound image LL on the left side of the end sound to be localized at the installation position of the left speaker 23SL, and the sound image RR on the right side of the end sound to be localized at the installation position of the right speaker 23SR.

[0117] Furthermore, the output control unit 103 causes the left speaker 23SL to output the Lch background sound generated based on the Lch uncorrelated component UL, and causes the right speaker 23SR to output the Rch background sound generated based on the Rch uncorrelated component UR. This allows the sound image UU of the background sound to be localized so as to extend in the space between the installation positions of the left speaker 23SL and the installation position of the right speaker 23SR.

[0118] Furthermore, the output control unit 103 causes the left main speaker 23ML to output the Lch middle sound generated based on the Lch-correlated components CC, CL1, CL2, CR1, and CR2, and causes the right main speaker 23MR to output the Rch middle sound generated based on the Rch-correlated components CC, CL1, CL2, CR1, and CR2. This allows the sound images C, L1, L2, R1, and R2 of the middle sound to be localized between the localized positions of the sound images LL and RR of the left and right end sounds.

[0119] At this time, the sound image localization adjustment unit 102 of this embodiment adjusts the left and right weights of the associated components CC, CL1, CL2, CR1, and CR2 so that the sound image L2 at the left end of the middle sound is positioned at the setting position of the left main speaker 23ML, and the sound image R2 at the right end is positioned at the setting position of the right main speaker 23MR.

[0120] FIG18 is a diagram illustrating an example of sound image localization adjustment processing performed on the associated components CC, CL1, CL2, CR1, and CR2 according to the first embodiment. FIG18 illustrates the state before adjustment in the upper portion, and the state after adjustment in the lower portion. In the state before adjustment, since the weight of the right channel of the associated component CL2 at the left end is greater than 0 (0.44 in this example), the sound image L2 at the left end of the center sound is localized inward relative to the installation position of the left main speaker 23ML. Similarly, since the weight of the left channel of the associated component CR2 at the right end is greater than 0 (0.44 in this example), the sound image R2 at the right end of the center sound is localized inward relative to the installation position of the right main speaker 23MR.

[0121] Therefore, as shown in the table at the bottom of Figure 18 , by setting the weight of the right channel of the left-side correlation component CL2, that is, the output level of the right main speaker 23MR, to 0, and by appropriately adjusting the other weights, the sound image L2 of the left end of the center sound can be localized at the installation position of the left main speaker 23ML. Similarly, by setting the weight of the left channel of the right-side correlation component CR2, that is, the output level of the left main speaker 23ML, to 0, and by appropriately adjusting the other weights, the sound image R2 of the right end of the center sound can be localized at the installation position of the right main speaker 23MR.

[0122] FIG19 is a flowchart illustrating an example of processing performed by the DSP 21 according to the first embodiment. In step S101, the separation unit 101 separates the independent components IL, IR, the uncorrelated components UL, UR, and the correlated components CC, CL1, CL2, CR1, and CR2 from the input signals il and ir. Specifically, the separation unit 101 separates the Lch independent component IL, the Lch uncorrelated component UL, and the Lch correlated components CC, CL1, CL2, CR1, and CR2 from the Lch input signal il, and separates the Rch independent component IR, the Rch uncorrelated component UR, and the Rch correlated components CC, CL1, CL2, CR1, and CR2 from the Rch input signal ir.

[0123] In step S102, the sound image localization adjustment unit 102 performs sound image localization adjustment processing on the independent components and the correlated components so that the left and right sound images LL and RR of the end sounds are respectively localized at the installation positions of the left and right speakers 23SL and 23R (an example of the left and right ends of the display device 1), and the sound images C, L1, L2, R1, and R2 of the center sounds are localized between the localized positions of the left and right sound images LL and RR of the end sounds. At this time, for example, the Lch independent component IL is adjusted so that the output level of the R channel is 0, and the Rch independent component IR is adjusted so that the output level of the LR channel is 0. In addition, as shown in FIG18, the correlated components CC, CL1, CL2, CR1, and CR2 are adjusted so that the output level of the R channel of the left component is 0 and the output level of the L channel of the right component is 0.

[0124] In step S103, the output control unit 103 causes the left speaker 23SL and the right speaker 23SR to output the end sound generated based on the independent components and the background sound generated based on the uncorrelated components. As a result, the sound image LL on the left side of the end sound is localized at the location of the left speaker 23SL, and the sound image RR on the right side of the end sound is localized at the location of the right speaker 23SR. Furthermore, the sound image UU of the background sound is localized so as to extend in the space between the locations of the left and right speakers 23SL and 23SR.

[0125] Furthermore, in step S104, the output control unit 103 causes the main speakers 23ML and 23MR to output the intermediate sound generated based on the correlation components. This causes the sound images C, L1, L2, R1, and R2 of the intermediate sound to be localized between the left and right sound images LL and RR of the end sounds. It should be noted that steps S103 and S104 are actually executed in parallel.

[0126] As described above, according to this embodiment, the input signal is separated into independent, uncorrelated, and correlated components. The center sound generated by the correlated components is output from the main speakers 23ML and 23MR, while the end sounds generated by the independent components are output from the left speaker 23SL and the right speaker 23R, which are positioned outboard of the main speakers 23ML and 23MR. This allows the sound images LL and RR of the end sounds to be localized at the ends of the display 31, ensuring a wide area for localizing the sound images C, L1, L2, R1, and R2 of the center sound. This prevents degradation in sound quality or a reduction in immersion caused by overlapping of the sound images C, L1, L2, R1, and R2 of the center sound or the sound images LL and RR of the end sounds. Furthermore, by outputting the background sound generated by the uncorrelated components from the left speaker 23SL and the right speaker 23SR, the background sound image UU can be localized so that it extends across the entire left and right areas of the display 31. This creates a high-quality sound image and allows for the output of high-quality sound with a high sense of immersion.

[0127] Hereinafter, although other embodiments will be described with reference to the drawings, descriptions of parts identical or similar to those of the first embodiment will be appropriately omitted.

[0128] (Second embodiment)

[0129] FIG20 illustrates an example configuration of multiple speakers 23 in a display device 1 according to a second embodiment. In this embodiment, the left-side speaker 23SL and the right-side speaker 23SM of the first embodiment are not present. In other words, in this embodiment, all end sounds, background sounds, and intermediate sounds are output from the main speakers 23ML and 23MR.

[0130] Fig. 21 is a diagram showing an example of the functional configuration of the DSP 21 according to the second embodiment. Fig. 22 is a diagram showing an example of the relationship between the output position of each sound and the sound image according to the second embodiment.

[0131] 21 , the output control unit 103 of this embodiment causes the left main speaker 23ML to output the Lch end sound, Lch background sound, and Lch middle sound, and causes the right main speaker 23MR to output the Rch end sound, Rch background sound, and Rch middle sound.

[0132] Furthermore, as shown in FIG22 , the sound image localization adjustment unit 102 of this embodiment adjusts the independent components so that the left sound image LL of the end sound is localized to the left of the left main speaker 23ML, and the right sound image RR of the end sound is localized to the right of the right main speaker 23MR. Furthermore, the sound image localization adjustment unit 102 of this embodiment adjusts the uncorrelated components so that the left end of the sound image UU of the background sound is located to the left of the left main speaker 23ML, and the right end of the sound image UU is located to the right of the right main speaker 23MR. Furthermore, the sound image localization adjustment unit 102 of this embodiment adjusts the correlated components so that the left end of the sound image L2 of the center sound is localized to the left of the left main speaker 23ML, and the right end of the sound image R2 of the center sound is localized to the right of the right main speaker 23MR.

[0133] 23 is a diagram showing an example of sound image localization adjustment processing for independent components IL, IR and uncorrelated components UL, UR according to the second embodiment. The sound image localization adjustment unit 102 of this embodiment adjusts the independent components IL, IR and uncorrelated components UL, UR using crosstalk cancellation.

[0134] In FIG23 , the components indicated by dotted lines correspond to correction components for achieving crosstalk cancellation. That is, as components of Lch, a correction component -0.96*IR' of opposite phase is added to the independent component 1.96*IR of Rch, and a correction component -0.96*UR' of opposite phase is added to the uncorrelated component 1.96*UR of Rch. Furthermore, as components of Rch, a correction component -0.96*IL' of opposite phase is added to the independent component 1.96*IL of Lch, and a correction component -0.96*UL' of opposite phase is added to the uncorrelated component 1.96*UL of Lch. Such correction components can be generated using, for example, a well-known crosstalk cancellation circuit. By using such correction components, the sound image LL on the left side of the end sound can be moved to the left side of the setting position of the left main speaker 23ML, and the sound image RR on the right side of the end sound can be moved to the right side of the setting position of the right main speaker 23MR. Furthermore, the sound image UU of the background sound can be expanded so that its left end reaches to the left of the installation position of the left main speaker 23ML and its right end reaches to the right of the installation position of the right main speaker 23MR.

[0135] 24 is a flowchart showing an example of processing by the DSP 21 according to the second embodiment. In step S201, the separation unit 101 separates the input signals il and ir into independent components IL and IR, uncorrelated components UL and UR, and correlated components CC, CL1, CL2, CR1, and CR2.

[0136] In step S202, the sound image localization adjustment unit 102 performs sound field localization adjustment processing to adjust the independent components so that the left sound image LL of the end sound is localized to the left of the left main speaker 23ML, and the right sound image RR of the end sound is localized to the right of the right main speaker 23MR. Furthermore, the sound image localization adjustment unit 102 performs sound field localization adjustment processing to adjust the uncorrelated components so that the left end of the sound image UU of the background sound is located to the left of the left main speaker 23ML, and the right end of the sound image UU is located to the right of the right main speaker 23MR. Furthermore, the sound image localization adjustment unit 102 performs sound field localization adjustment processing to adjust the correlated components so that the left end of the sound image L2 of the center sound is localized to the left of the left main speaker 23ML, and the right end of the sound image R2 of the center sound is localized to the right of the right main speaker 23MR.

[0137] In step S203, the output control unit 103 causes the main speakers 23ML and 23MR to output the end sound generated by the independent components, the background sound generated by the uncorrelated components, and the center sound generated by the correlated components. As a result, the left and right sound images LL and RR of the end sound are positioned outward from the left and right main speakers 23ML and 23MR, respectively, and the sound images C, L1, L2, R1, and R2 of the center sound are positioned between the left and right sound images LL and RR of the end sound. Furthermore, the sound image UU of the background sound is positioned so that its left end is positioned to the left of the position of the left main speaker 23ML and its right end is positioned to the right of the position of the right main speaker 23MR.

[0138] As described above, according to this embodiment, the sound images LL and RR of the end sounds generated by the independent components can be localized near the left and right ends of the display 31 without using the left and right speakers 23SL and 23SM, thereby ensuring a wide area for localizing the sound images C, L1, L2, R1, and R2 of the center sounds. Furthermore, the sound image UU of the background sound can be localized so that it extends across the entire left and right area of ​​the display 31. This allows for the production of high-quality sound images using two speakers while minimizing cost increases and providing high-quality, immersive sound output.

[0139] The program for realizing the functions of the display device 1 (DSP 21) described above is provided as a file in a format installable in a computer or in an executable format, stored on a computer-readable storage medium such as a CD-ROM, a floppy disk (FD), a CD-R, or a DVD (Digital Versatile Disk). Alternatively, the program may be stored on a computer connected to a network such as the Internet and provided by downloading the program via the network. Alternatively, the program may be provided or distributed via a network such as the Internet.

[0140] Although the embodiments of the present application have been described above, these embodiments are provided as examples and are not intended to limit the scope of the application. These new embodiments may be implemented in various other ways, and various omissions, substitutions, and modifications may be made without departing from the gist of the invention. These embodiments or modifications thereof are included within the scope or gist of the invention and are included within the scope equivalent to the invention described in the claims.

Claims

1. A sound field correction device that performs processing for reducing the deviation of the timing at which sounds outputted from a plurality of speakers reach a predetermined listening position, wherein: The sound field correction device comprises: a test sound output unit that outputs a first test sound of a first frequency from a first speaker and a second speaker, outputs a second test sound of a second frequency different from the first frequency from the first speaker when a first standby time has passed after the first test sound is output, and outputs the second test sound from the second speaker when a second standby time different from the first standby time has passed after the first test sound is output; a recording unit that collects sound through a microphone disposed at the listening position and records a composite sound, wherein the composite sound includes the first test sound and the second test sound outputted from the first speaker and the second speaker respectively; a calculation unit, which calculates a first time difference and a second time difference based on a result of a frequency analysis process, wherein the frequency analysis process is a process capable of detecting a component corresponding to the first frequency and a component corresponding to the second frequency from the composite sound of the recorded sound, the first time difference being a time difference between a timing when the first test sound output by the first speaker reaches the microphone and a timing when the first test sound output by the second speaker reaches the microphone, and the second time difference being a time difference between a timing when the second test sound output by the first speaker reaches the microphone and a timing when the second test sound output by the second speaker reaches the microphone; as well as A correction unit corrects at least one of an output timing of the sound from the first speaker or an output timing of the sound from the second speaker based on the first time difference and the second time difference.

2. The sound field correction device according to claim 1, wherein: The correction unit determines the order of when the first test sound output by the first speaker arrives at the microphone and when the first test sound output by the second speaker arrives at the microphone based on the size relationship between the first time difference and the second time difference, and based on the order of when the first test sound output by the first speaker arrives at the microphone, relatively delays either the output timing of the sound from the first speaker or the output timing of the sound from the second speaker to reduce the first time difference.

3. A sound field correction method for reducing the deviation in timing when sounds outputted from a plurality of speakers reach a predetermined listening position, wherein: The sound field correction method comprises: The step of outputting a first test tone of a first frequency from the first speaker and the second speaker; a step of outputting a second test tone having a second frequency different from the first frequency from the first speaker when a first waiting time has elapsed after the first test tone is output; a step of outputting the second test tone from the second speaker when a second standby time different from the first standby time has elapsed after the first test tone is output; A step of collecting sound by a microphone disposed at the listening position to record a composite sound, wherein the composite sound includes the first test sound and the second test sound outputted from the first speaker and the second speaker respectively; The process of calculating a first time difference and a second time difference based on a result of a frequency analysis process, wherein the frequency analysis process is a process capable of detecting a component corresponding to the first frequency and a component corresponding to the second frequency from the composite sound recorded, wherein the first time difference is a time from when the first test sound output by the first speaker reaches the microphone The second time difference is the time difference between the timing when the second test tone output by the first speaker arrives at the microphone and the timing when the first test tone output by the second speaker arrives at the microphone; as well as A step of correcting at least one of a timing of outputting sound from the first speaker or a timing of outputting sound from the second speaker based on the first time difference and the second time difference.

4. A non-volatile storage medium storing a program, wherein: The program causes an information processing device that performs processing for reducing variations in timings at which sounds output from a plurality of speakers reach a predetermined listening position to execute the following processing: outputting a first test tone of a first frequency from the first speaker and the second speaker; When a first standby time has passed after the first test tone is output, a second test tone having a second frequency different from the first frequency is output from the first speaker; After the first test tone is output, when a second standby time different from the first standby time has passed, outputting the second test tone from the second speaker; Collecting sound by a microphone disposed at the listening position, recording a composite sound, wherein the composite sound includes the first test sound and the second test sound outputted from the first speaker and the second speaker respectively; calculating a first time difference and a second time difference based on a result of a frequency analysis process, wherein the frequency analysis process is a process capable of detecting a component corresponding to the first frequency and a component corresponding to the second frequency based on the composite sound of the recorded sound, the first time difference is a time difference between a timing when the first test sound output by the first speaker reaches the microphone and a timing when the first test sound output by the second speaker reaches the microphone, and the second time difference is a time difference between a timing when the second test sound output by the first speaker reaches the microphone and a timing when the second test sound output by the second speaker reaches the microphone; as well as At least one of the output timing of the sound from the first speaker or the output timing of the sound from the second speaker is corrected based on the first time difference and the second time difference.

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