CN116803104A - Sound field correction device, sound field correction method, and nonvolatile storage medium - Google Patents

Abstract

Provided are a sound field correction device, a sound field correction method, and a nonvolatile storage medium. The sound field correction apparatus effectively suppresses an influence due to an arrival time difference of sound between the plurality of speakers. The sound field correction device is provided with: a test sound output unit that simultaneously outputs test sounds having different frequencies for each speaker, respectively; a recording unit for recording a sound signal of a composite sound including a plurality of test sounds collected by a microphone disposed at a predetermined position; a calculation unit that calculates a time difference in arrival time of a test sound between a plurality of speakers from the speaker to the microphone, based on analysis results of a plurality of frequency components included in the sound signal; and a correction unit that corrects the output timing of the sound output from each speaker based on the time difference.

Description

Sound field correction device, sound field correction method, and nonvolatile storage medium

Cross Reference to Related Applications

The present application claims priority from japanese patent office, application No. 2022-044965, japanese patent application entitled "sound field correction apparatus, sound field correction method, and program" filed on 3 month 22 of 2022, the entire contents of which are incorporated herein by reference.

Technical Field

Embodiments of the present application relate to a sound field correction apparatus, a sound field correction method, and a nonvolatile storage medium.

Background

In some acoustic devices that output sound from a plurality of speakers, the arrival time of the sound from the speaker to the listening position may be different for each speaker. As a method for solving such a problem, there is a technique as follows: the test sound output from the speaker is collected by the microphone, the distance between the speaker and the microphone is calculated based on the output signal of the microphone, and the output timing of the sound is adjusted (delayed) according to the distance.

Prior art literature

Patent literature

Patent document 1: japanese patent laid-open No. 2007-110357

Disclosure of Invention

The above-described difference in arrival time of the sound is generated due to a difference in distance between each speaker and the listening position, a difference in communication speed caused by the intervention of wireless communication, or the like, but in the related art, there are cases where the difference in arrival time of the sound cannot be sufficiently eliminated.

An object of an embodiment of the present application is to provide a sound field correction device, a sound field correction method, and a nonvolatile storage medium that can effectively suppress an influence due to a difference in arrival time of sound between a plurality of speakers.

The sound field correction device of the embodiment is provided with: a test sound output unit that simultaneously outputs test sounds having different frequencies for each speaker, respectively; a recording unit for recording a sound signal of a composite sound including a plurality of test sounds collected by a microphone disposed at a predetermined position; a calculation unit that calculates a time difference in arrival time of a test sound between a plurality of speakers from the speaker to the microphone, based on analysis results of a plurality of frequency components included in the sound signal; and a correction unit that corrects the output timing of the sound output from each speaker based on the time difference.

Drawings

Fig. 1 is a block diagram showing an example of the hardware configuration of the acoustic apparatus according to embodiment 1;

fig. 2 is a block diagram showing an example of the functional configuration of the sound field correction apparatus of embodiment 1;

fig. 3 is a diagram showing an example of the positional relationship among the 1 st speaker, the 2 nd speaker, and the microphone according to embodiment 1;

fig. 4 is a diagram showing an example of a sound signal of a composite sound according to embodiment 1;

fig. 5 is a diagram showing an example of the result of FFT processing of an acoustic signal according to embodiment 1;

fig. 6 is a diagram showing an example of the extraction result of the 1 st frequency and the extraction result of the 2 nd frequency in embodiment 1;

fig. 7 is a flowchart showing an example of processing in the sound field correction apparatus of embodiment 1;

fig. 8 is a block diagram showing an example of the hardware configuration of the audio apparatus according to embodiment 2;

fig. 9 is a block diagram showing an example of the functional configuration of the sound field correction apparatus of embodiment 2;

fig. 10 is a flowchart showing example 1 of the process in the sound field correction apparatus of embodiment 2;

fig. 11 is a flowchart showing an example 2 of the processing in the sound field correction apparatus of embodiment 2.

Description of the reference numerals

1. 2 … sound device; 5. 6 … sound field correction means; 11 … CPU;12 … memory; 13 … memory; 14 … user I/F;15 … communication I/F;21 … sound decoder; 22 … sound input ADC;23 … DSP;25a … 1 st delay circuit; 25B … delay circuit 2; 25C … 3 rd delay circuit; 31a … 1 st speaker; 31B … speaker 2; 38a … 1 st internal speaker; 38B … 2 nd internal speaker; 41 … remote control; 45 … microphone; 46 … wireless modulation circuitry; 47 … transmitter; 101 … sound output unit; 102 … test tone output section; 103 … trusted section; 104 … recording part; 105 and … arithmetic unit; 106 … correction unit; AS1, AS2 … sounds; f … analyzing the result; extracting results of F1 and F2 …; r … subject frequency domain; a St … sound signal; t1, T2 … arrival times; TS1 … test tone 1; TS2 …, test tone 2; TS3 … 3 rd test tone; w … time window; Δt … time difference; Δtint … internal time difference; Δtext … external time difference.

Detailed Description

Illustrative embodiments of the application are disclosed below.

Embodiment 1

Fig. 1 is a block diagram showing an example of the hardware configuration of the acoustic apparatus 1 according to embodiment 1. The audio device 1 is a device capable of outputting sound from a plurality of speakers, and may be, for example, a stereo, video playback device, video recording device, television, home theater system, or the like.

The acoustic device 1 of the present embodiment includes a sound field correction device 5, a 1 st speaker 31A, a 2 nd speaker 31B, and a remote controller 41. The sound field correction apparatus 5 performs the following sound field correction processing: the output timings of the sounds AS1 and AS2 output from the 1 st speaker 31A and the 2 nd speaker 31B are optimized based on the positional relationship between the 1 st speaker 31A, the 2 nd speaker 31B, and the listening position (Listening position) of the user. The listening position in the present embodiment is set to a position where the remote control 41 is present.

The sound field correction apparatus 5 of the present embodiment includes a CPU (Central Processing unit: central processing unit) 11, a memory 12, a storage 13, a user I/F (Interface) 14, a communication I/F15, a sound decoder 21, a sound input ADC (Analog to Digital Converter: analog-digital converter) 22, a DSP (Digital Signal Processor: digital signal processor) 23, a 1 st delay circuit 25A, and a 2 nd delay circuit 25B, which are communicably connected to each other via a communication bus 20.

The CPU11 executes predetermined arithmetic processing and control processing in accordance with programs (including firmware, application software, and the like) stored in the memory 12 and the like. The Memory 12 is a main storage device including a RAM (Random Access Memory: random access Memory), a ROM (Read Only Memory), and the like, and functions as a program storage area, a work area of the CPU11, and the like. The memory 13 is an auxiliary storage device including a nonvolatile memory such as an SSD (Solid State Drive) or an HDD (Hard Disk Drive), and can write and read various data. The user I/F14 is a device capable of receiving input from a user, outputting information to the user, and the like, and may be a display, an input button, or the like, for example. The communication I/F15 is a device capable of communication with other electronic devices connected via a predetermined communication network. Wireless communication according to a predetermined standard is established between the communication I/F15 of the present embodiment and the remote controller 41.

The audio decoder 21 is a device for converting audio data recorded on a predetermined medium (for example, a CD, DVD, blu-ray (registered trademark) disc, mobile medium, or the like), audio data included in a broadcast wave, audio data acquired from a network such as CSP (Communications Service Provider: communication service provider), or the like, into digital signals in a form that can be output from the 1 st speaker 31A and the 2 nd speaker 31B. The sound input ADC22 is a device that converts an analog signal of sound input from an external device into a digital signal.

The DSP23 is a processor that performs predetermined processing on digital signals corresponding to the sounds output from the 1 st speaker 31A and the 2 nd speaker 31B, and generates sound signals of the sounds AS1 and AS2 to be listened to, sound signals of the 1 st test sound TS1 and the 2 nd test sound TS2 described later, and the like. The 1 st delay circuit 25A is a circuit that delays the output timing of the sound AS1 output from the 1 st speaker 31A (for example, lch speaker) based on the correction signal (delay signal) output from the CPU 11. The 2 nd delay circuit 25B is a circuit that delays the output timing of the sound AS2 output from the 2 nd speaker 31B (for example, rch speaker) according to the correction signal output from the CPU 11.

The remote control 41 is a device operable by a user who wants 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 collected sound into an electrical signal (analog signal). The wireless modulation circuit 46 is a circuit that modulates an electric signal generated by the microphone 45 into a signal (digital signal) in a form that enables wireless communication according to a predetermined standard. The transmitter 47 is a device that transmits the signal modulated by the wireless modulation circuit 46 to the sound field correction apparatus 5. The remote controller 41 is provided with buttons or the like for receiving operations performed by the user in addition to the above-described devices, but the description thereof is omitted here. In the present embodiment, the microphone 45 is provided in the remote controller 41, but the microphone 45 may be a separate device.

When performing the sound field correction process, the sound field correction apparatus 5 of the present embodiment outputs the 1 st test sound TS1 and the 2 nd test sound TS2 from the 1 st speaker 31A and the 2 nd speaker 31B, respectively. The frequency of the 1 st test tone TS1 and the frequency of the 2 nd test tone TS2 are different from each other. The microphone 45 mounted on the remote controller 41 collects a composite sound including the 1 St test sound TS1 and the 2 nd test sound TS2, and the transmitter 47 transmits the sound signal St of the composite sound collected by the microphone 45 to the sound field correction device 5. The sound field correction apparatus 5 calculates a time difference between the arrival time from the 1 St speaker 31A of the 1 St test tone TS1 to the microphone 45 and the arrival time from the 2 nd speaker 31B of the 2 nd test tone TS2 to the microphone 45 based on the analysis result of the frequency component contained in the sound signal St received from the remote controller 41. Then, the sound field correction device 5 performs a process of correcting at least one of the output timing of the sound AS1 from the 1 st speaker 31A and the output timing of the sound AS2 from the 2 nd speaker 31B, that is, a process for controlling the 1 st delay circuit 25A or the 2 nd delay circuit 25B, based on the calculated time difference.

Fig. 2 is a block diagram showing an example of the functional configuration of sound field correction apparatus 5 according to embodiment 1. The sound field correction apparatus 5 of the present embodiment includes a sound output unit 101, a test sound output unit 102, a receiving unit 103, a recording unit 104, a computing unit 105, and a correction unit 106. The constituent elements 101 to 106 of these functions can be realized by cooperation of hardware and software (programs) as illustrated in fig. 1. At least some of the functional components 101 to 106 may be realized by dedicated hardware (circuits and the like).

The audio output unit 101 outputs audio AS1 and AS2 to be listened to from the 1 st speaker 31A and the 2 nd speaker 31B, respectively.

When performing the sound field correction process, the test tone output unit 102 outputs the 1 st test tone TS1 from the 1 st speaker 31A and the 2 nd test tone TS2 from the 2 nd speaker 31B. The 1 st test tone TS1 and the 2 nd test tone TS2 are different in frequency, and the 1 st test tone TS1 and the 2 nd test tone TS2 are simultaneously output. The test tone output unit 102 may repeat the test tone group including the 1 st test tone TS1 and the 2 nd test tone TS2 which are simultaneously output at predetermined time intervals. For example, the test tone group including the 1 st test tone TS1 and the 2 nd test tone TS2 which are simultaneously output may be output a plurality of times (for example, 5 times or the like) at predetermined time intervals during the execution period of the sound field correction process. This can increase information used in the frequency analysis processing described later, and can improve the accuracy of the sound field correction processing.

The receiving unit 103 receives the audio signal St of the composite sound including the 1 St test sound TS1 and the 2 nd test sound TS2. The receiving unit 103 of the present embodiment receives the audio signal St from the remote controller 41 via wireless communication. The receiving unit 103 may receive the audio signal St via wired communication.

The recording unit 104 stores the audio signal St received by the receiving unit 103 in a predetermined storage device (for example, the memory 13).

The arithmetic unit 105 performs a frequency analysis process for analyzing a plurality of frequency components included in the audio signal St, on the audio signal St recorded by the recording unit 104. Then, the calculation unit 105 calculates a time difference Δt between the arrival time of the 1 st speaker 31A of the 1 st test tone TS1 to the microphone 45 and the arrival time of the 2 nd speaker 31B of the 2 nd test tone TS2 to the microphone 45 based on the analysis result of the frequency analysis process. When the test tone sets are output repeatedly a plurality of times, the arithmetic unit 105 performs the frequency analysis process described above for each test tone set. This can improve the accuracy of the time difference Δt.

The correction unit 106 corrects the output timings of the sounds AS1 and AS2 output from the 1 st speaker 31A and the 2 nd speaker 31B, respectively, based on the time difference Δt calculated by the calculation unit 105. For example, when the arrival time of the 1 st test tone TS1 is later than the arrival time of the 2 nd test tone TS2 by Δt, the output time of the sound AS2 is delayed by Δt from the output time of the sound AS 1. In addition, when the test tone groups are repeatedly output a plurality of times, the correction unit 106 corrects the output timing based on a plurality of time differences Δt calculated for each test tone group (for example, an average value of a plurality of time differences Δt, etc.). This can improve the accuracy of correction of the output timing.

With the above configuration, the arrival time differences of the respective sounds AS1 and AS2 due to the positional relationship between the plurality of speakers (1 st speaker 31A and 2 nd speaker 31B in the present embodiment) and the listening position, communication delay caused by the intervention of wireless communication, and the like can be corrected with high accuracy.

Fig. 3 is a diagram showing an example of the positional relationship of the 1 st speaker 31A, the 2 nd speaker 31B, and the microphone 45 according to embodiment 1. In fig. 3, a case is illustrated in which the distance D1 between the 1 st speaker 31A and the microphone 45 is greater than the distance D2 between the 2 nd speaker 31B and the microphone 45. In addition, it is shown that the 1 st frequency Ft1 of the 1 st test tone TS1 output from the 1 st speaker 31A and the 2 nd frequency Ft2 of the 2 nd test tone TS2 output from the 2 nd speaker 31B are different from each other. Here, although the case where Ft1 < Ft2 is illustrated, ft1 > Ft2 may be used.

In the above-described situation, if the 1 st test tone TS1 and the 2 nd test tone TS2 are simultaneously output from the 1 st speaker 31A and the 2 nd speaker 31B, respectively, the 2 nd test tone TS2 reaches the microphone 45 earlier than the 1 st test tone TS 1. Thus, in the sound signal St of the composite sound collected by the microphone 45, there are a period of time including the component of the 2 nd frequency Ft2 and not including the component of the 1 St frequency Ft1, and a period of time including the components of both the 1 St frequency Ft1 and the 2 nd frequency Ft2. Then, the frequency component included in such a sound signal St can be analyzed by an appropriate method such as FFT (Fast Fourier Transform: fast fourier transform), and the time difference Δt can be calculated.

Fig. 4 is a diagram showing an example of the sound signal St of the composite sound of embodiment 1. In the graph shown in fig. 4, the horizontal axis represents the elapsed time from the predetermined reference time (for example, the output time of the 1 st test tone TS1 and the 2 nd test tone TS2, the recording start time of the composite tone, etc.), and the vertical axis represents the signal intensity of the composite tone acquired by the microphone 45. In fig. 4, the arrival time T1 corresponds to the time when the 1 st test tone TS1 arrives at the microphone 45 from the 1 st speaker 31A, and the arrival time T2 corresponds to the time when the 2 nd test tone TS2 arrives at the microphone 45 from the 2 nd speaker 31B. The time difference Δt is the difference between the arrival time T1 and the arrival time T2.

The calculation of the time difference Δt described above may be performed using an appropriate method. Hereinafter, an example of calculating the time difference Δt using FFT is shown.

Fig. 5 is a diagram showing an example of the result of FFT processing of the audio signal St according to embodiment 1. Fig. 5 illustrates the analysis result F obtained when FFT processing is performed for the time window W of the predetermined time frame amount set in the audio signal St.

In the analysis result F, the horizontal axis corresponds to frequency and the vertical axis corresponds to power spectrum. In the present embodiment, a power value corresponding to the predetermined target frequency domain R is calculated from the analysis result F. The target frequency domain R is set with reference to the 1 st frequency Ft1 at the time of extraction of the 1 st frequency Ft1, and with reference to the 2 nd frequency Ft2 at the time of detection of the 2 nd frequency Ft2. For example, the target frequency domain R at the time of extraction of the 1 st frequency Ft1 is set to (ft1±x%) and is, for example, r=500±50 (Hz) when ft1=500 (Hz) and x=10. Similarly, the target frequency domain R at the time of extraction of the 2 nd frequency Ft2 is set to (ft2±x%) such that, for example, r=2000±200 (Hz) in the case of ft2=2000 (Hz) and x=10. By the above-described processing, the power value corresponding to the 1 st frequency Ft1 (for example, r=500±50) and the power value corresponding to the 2 nd frequency Ft2 (r=2000±200) are obtained for each of the plurality of time windows W gradually shifted at predetermined time intervals.

Fig. 6 is a diagram showing an example of the extraction results F1 and F2 of the 1 st frequency Ft1 and the 2 nd frequency Ft2 of embodiment 1. In the extraction result F1, the horizontal axis corresponds to time, and the vertical axis corresponds to a power value corresponding to the target frequency domain R (e.g., 500±50) of the 1 st frequency Ft 1. In the extraction result F2, the horizontal axis corresponds to time, and the vertical axis corresponds to a power value corresponding to the target frequency domain R (e.g., 2000±200) of the 2 nd frequency Ft2. The time axis of the extraction result F1 coincides with the time axis of the extraction result F2. In the extraction result F1, the time when the power value reaches the threshold Th1 corresponds to the arrival time T1 of the 1 st test tone TS1 from the 1 st speaker 31A to the microphone 45. In the extraction result F2, the time when the power value reaches the threshold Th2 corresponds to the arrival time T2 of the 2 nd test tone TS2 from the 2 nd speaker 31B to the microphone 45. Then, the difference between the arrival time T1 and the arrival time T2 thus obtained becomes the above-described time difference Δt.

Fig. 7 is a flowchart showing an example of processing in the sound field correction apparatus 5 of embodiment 1. When the sound field correction process starts, the test sound output section 102 outputs the 1 st test sound TS1 from the 1 st speaker 31A and the 2 nd test sound TS2 from the 2 nd speaker 31B (S101). At this time, the 1 st test tone TS1 and the 2 nd test tone TS2 are simultaneously output. The composite sound including the 1 St test sound TS1 and the 2 nd test sound TS2 is collected by the microphone 45 disposed at the listening position, and the sound signal St of the composite sound is recorded by the recording unit 104 (S102).

Thereafter, the arithmetic unit 105 extracts the 1 St frequency component corresponding to the 1 St test tone TS1 and the 2 nd frequency component corresponding to the 2 nd test tone TS2 from the audio signal St by using a method such as FFT (S103). The calculation unit 105 calculates an arrival time T1 of the 1 st speaker 31A (a time when the 1 st test sound TS1 arrives at the microphone 45 from the 1 st speaker 31A) and an arrival time T2 of the 2 nd speaker 31B (a time when the 2 nd test sound TS2 arrives at the microphone 45 from the 2 nd speaker 31B) (S104). The computing unit 105 calculates a time difference Δt from the difference between the arrival time T1 and the arrival time T2 (S105). Then, the correction unit 106 controls the 1 st delay circuit 25A and the 2 nd delay circuit 25B based on the time difference Δt so AS to optimize the output timing of the sound AS1 from the 1 st speaker 31A and the output timing of the sound AS2 from the 2 nd speaker 31B (reduce the time difference Δt) (S106).

According to the above embodiment, the time difference Δt is calculated based on the analysis result of the frequency components included in the recorded composite sound, that is, the comparison result of the time when each frequency component is detected in the recorded sound signal St. This can eliminate the influence of the time from the instruction to output the test sound to the actual output of the test sound from the speaker, the time from the acquisition of the test sound by the microphone to the recording, and the like. Thus, the influence due to the difference in arrival time of the sound between the plurality of speakers to the listening position can be effectively suppressed, and a comfortable sound field can be provided.

Hereinafter, other embodiments will be described, and the description thereof will be omitted as appropriate for those parts that exhibit the same or similar effects as those of embodiment 1.

Embodiment 2

Fig. 8 is a block diagram showing an example of the hardware configuration of the audio apparatus 2 according to embodiment 2. In the present embodiment, the acoustic device 2 includes the 1 st internal speaker 38A and the 2 nd internal speaker 38B, and the external speaker 51 is connected to the acoustic device 2. The 1 st internal speaker 38A and the 2 nd internal speaker 38B are speakers mounted in a predetermined case (for example, a case constituting a main body of the acoustic device 2). The external speaker 51 is, for example, a speaker disposed outside a cabinet on which the 1 st internal speaker 38A and the 2 nd internal speaker 38B are mounted.

The sound field correction apparatus 6 of the present embodiment simultaneously outputs the 1 st test sound TS1, the 2 nd test sound TS2, and the 3 rd test sound TS3 from the 1 st internal speaker 38A, the 2 nd internal speaker 38B, and the external speaker 51, respectively. The sound field correction device 6 performs sound field correction processing that optimizes output timings of the sounds AS1, AS2, AS3 output from the 1 st internal speaker 38A, the 2 nd internal speaker 38B, and the external speaker 51, respectively, based on arrival time differences of the 1 st test tone TS1, the 2 nd test tone TS2, and the 3 rd test tone TS3 reaching the microphone 45, respectively.

The acoustic device 2 of the present embodiment includes a transmitter 27 that establishes wireless communication with the external speaker 51. The sound field correction apparatus 6 of the present embodiment includes a 3 rd delay circuit 25C connected to the transmitter 27. By the action of the 3 rd delay circuit 25C, the output timing of the sound AS3 from the external speaker 51 can be adjusted. In the present embodiment, the configuration in which the acoustic device 2 and the external speaker 51 are connected via wireless communication is exemplified, but the acoustic device 2 and the external speaker 51 may be connected via wired communication.

Fig. 9 is a block diagram showing an example of the functional configuration of sound field correction apparatus 6 according to embodiment 2. The audio output unit 101 of the present embodiment causes the 1 st internal speaker 38A, the 2 nd internal speaker 38B, and the external speaker 51 to output the audio AS1, AS2, AS3 to be listened to, respectively. In addition, when performing the sound field correction processing, the sound output unit 101 of the present embodiment causes the 1 st internal speaker 38A, the 2 nd internal speaker 38B, and the external speaker 51 to simultaneously output the 1 st test sound TS1, the 2 nd test sound TS2, and the 3 rd test sound TS3, respectively.

The recording unit 104 of the present embodiment records the sound signal St of the composite sound including at least 2 out of the 1 St test sound TS1, the 2 nd test sound TS2, and the 3 rd test sound TS3. The arithmetic unit 105 of the present embodiment calculates an internal time difference Δtint, which is a time difference between the arrival time T1 of the 1 St test tone TS1 and the arrival time T2 of the 2 nd test tone TS2, and an external time difference Δtext, which is a time difference between the arrival time (for example, the arrival time T1) of at least one of the 1 St test tone TS1 and the 2 nd test tone TS2 and the arrival time T3 of the 3 rd test tone TS3, based on an analysis result of the frequency analysis process of the audio signal St. The correction unit 106 of the present embodiment corrects the output timing of at least one of the 1 st internal speaker 38A and the 2 nd internal speaker 38B based on the internal time difference Δtint, as in the 1 st embodiment. The correction unit 106 of the present embodiment corrects the output time of at least 1 of the 1 st internal speaker 38A, the 2 nd internal speaker 38B, and the external speaker 51 based on the external time difference Δtext.

Fig. 10 is a flowchart showing example 1 of the processing in sound field correction apparatus 6 of embodiment 2. In this example, sound field correction processing is performed using 2 kinds of test tones.

When the sound field correction process is started, the test sound output section 102 outputs the 1 st test sound TS1 from the 1 st internal speaker 38A and the 2 nd test sound TS2 from the 2 nd internal speaker 38B (S201). At this time, the 1 st test tone TS1 and the 2 nd test tone TS2 are simultaneously output. The composite sound including the 1 St test sound TS1 and the 2 nd test sound TS2 is collected by the microphone 45 disposed at the listening position, and the sound signal St of the composite sound is recorded by the recording unit 104 (S202).

Thereafter, the arithmetic unit 105 extracts the 1 St frequency component corresponding to the 1 St test tone TS1 and the 2 nd frequency component corresponding to the 2 nd test tone TS2 from the audio signal St by using a method such as FFT (S203). The calculation unit 105 calculates the arrival time T1 of the 1 st internal speaker 38A (the time when the 1 st test sound TS1 arrives at the microphone 45 from the 1 st internal speaker 38A) and the arrival time T2 of the 2 nd internal speaker 38B (the time when the 2 nd test sound TS2 arrives at the microphone 45 from the 2 nd internal speaker 38B) (S204). The arithmetic unit 105 calculates an internal time difference Δtint from the difference between the arrival time T1 and the arrival time T2 (S205).

Thereafter, the test tone output unit 102 outputs the 1 st test tone TS1 from the 1 st internal speaker 38A and the 2 nd test tone TS2 as the 3 rd test tone TS3 from the external speaker 51 (S206). The composite sound including the 1 St test sound TS1 and the 2 nd test sound TS2 is collected by the microphone 45 disposed at the listening position, and the sound signal St of the composite sound is recorded in the memory 13 or the like by the recording unit 104 (S207).

Thereafter, the arithmetic unit 105 extracts the 1 St frequency component corresponding to the 1 St test tone TS1 and the 2 nd frequency component corresponding to the 2 nd test tone TS2 from the audio signal St by using a method such as FFT (S208). The calculation unit 105 calculates the arrival time T1 of the 1 st internal speaker 38A (the time when the 1 st test tone TS1 arrives at the microphone 45 from the 1 st internal speaker 38A) and the arrival time T3 of the external speaker 51 (the time when the 2 nd test tone TS2 arrives at the microphone 45 from the external speaker 51) (S209). The computing unit 105 calculates an external time difference Δtext from the difference between the arrival time T1 and the arrival time T3 (S210).

Then, the correction unit 106 controls the 1 st delay circuit 25A, the 2 nd delay circuit 25B, and the 3 rd delay circuit 25C based on the internal time difference Δtint and the external time difference Δtext so AS to optimize the output timing of the sound AS1 from the 1 st internal speaker 38A, the output timing of the sound AS2 from the 2 nd internal speaker 38B, and the output timing of the sound AS3 from the external speaker 51 (S211).

According to the above processing, the sound field correction processing for the sound field including 3 speakers (the 1 st internal speaker 38A, the 2 nd internal speaker 38B, and the external speaker 51) can be performed using 2 kinds of test sounds (the 1 st test sound TS1 and the 2 nd test sound TS 2). Thereby, the frequency analysis processing can be performed with a relatively simple algorithm.

Fig. 11 is a flowchart showing an example 2 of the processing in the sound field correction apparatus 6 of embodiment 2. In this example, sound field correction processing is performed using 3 kinds of test tones.

When the sound field correction process is started, the test sound output section 102 outputs the 1 st test sound TS1 from the 1 st internal speaker 38A, the 2 nd test sound TS2 from the 2 nd internal speaker 38B, and the 3 rd test sound TS3 from the external speaker 51 (S301). At this time, the 1 st test tone TS1, the 2 nd test tone TS2, and the 3 rd test tone TS3 are simultaneously output. The composite sound including the 1 St test sound TS1, the 2 nd test sound TS2, and the 3 rd test sound TS3 is collected by the microphone 45 disposed at the listening position, and the sound signal St of the composite sound is recorded by the recording unit 104 (S302).

Thereafter, the arithmetic unit 105 extracts the 1 St frequency component corresponding to the 1 St test tone TS1, the 2 nd frequency component corresponding to the 2 nd test tone TS2, and the 3 rd frequency component corresponding to the 3 rd test tone TS3 from the audio signal St by using a method such as FFT (S303). The calculation unit 105 calculates the arrival time T1 of the 1 st internal speaker 38A (the time when the 1 st test tone TS1 arrives at the microphone 45 from the 1 st internal speaker 38A), the arrival time T2 of the 2 nd internal speaker 38B (the time when the 2 nd test tone TS2 arrives at the microphone 45 from the 2 nd internal speaker 38B), and the arrival time T3 of the external speaker 51 (the time when the 3 rd test tone TS3 arrives at the microphone 45 from the external speaker 51) (S304). The computing unit 105 calculates an internal time difference Δtint from the difference between the arrival time T1 and the arrival time T2, and calculates an external time difference Δtext from the difference between the arrival time T1 and the arrival time T3 (S305). The external time difference Δtext may be calculated from the difference between the arrival time T2 and the arrival time T3.

Then, the correction unit 106 controls the 1 st delay circuit 25A, the 2 nd delay circuit 25B, and the 3 rd delay circuit 25C based on the internal time difference Δtint and the external time difference Δtext so AS to optimize the output timing of the sound AS1 from the 1 st internal speaker 38A, the output timing of the sound AS2 from the 2 nd internal speaker 38B, and the output timing of the sound AS3 from the external speaker 51 (S306).

According to the above processing, it is possible to perform sound field correction processing for sound fields of 3 speakers (1 st internal speaker 38A, 2 nd internal speaker 38B, and external speaker 51) including 3 types of test tones (1 st test tone TS1, 2 nd test tone TS2, and 3 rd test tone TS 3) that are simultaneously output. Thus, the sound field correction processing can be performed promptly.

The program for realizing the functions of the sound field correction apparatuses 5 and 6 described above may be provided by storing a file in a form that can be installed on a computer or in a form that can be executed by a computer in a storage medium readable by a computer, such as a CD-ROM, a Flexible Disk (FD), and a CD-R, DVD (Digital Versatile Disk: digital versatile disk). The program may be stored in a computer connected to a network such as the internet, and may be downloaded via the network. The program may be provided or distributed via a network such as the internet.

Embodiments of the present application also provide a computer-readable non-volatile storage medium having stored thereon computer instructions which, when executed by a computer, implement the method described in the above embodiments.

While the embodiments of the present application have been described above, these embodiments are presented as examples and are not intended to limit the scope of the application. These novel embodiments can be implemented in various other forms, and various omissions, substitutions, and changes can be made without departing from the spirit of the application. These embodiments and modifications thereof are included in the scope and spirit of the application, and are included in the claims and their equivalents.

Claims (6)

1. A sound field correction device is provided with:

a test sound output unit that simultaneously outputs test sounds having different frequencies for each of the plurality of speakers;

a recording unit that records a sound signal of a composite sound including a plurality of test sounds collected by a microphone disposed at a predetermined position;

a calculation unit that calculates a time difference in arrival time of the test sound from the speaker to the microphone between the plurality of speakers, based on analysis results of a plurality of frequency components included in the sound signal; and

and a correction unit that corrects an output timing of the sound output from each of the speakers based on the time difference.

2. The sound field correction apparatus as claimed in claim 1, wherein,

the test sound output unit repeatedly outputs a test sound group including a plurality of test sounds simultaneously output at predetermined time intervals,

the correction unit corrects the output timing based on the plurality of time differences calculated for each of the test tone groups.

3. The sound field correction apparatus as claimed in claim 1, wherein,

the sound field correction device further includes a receiving unit that receives the sound signal of the composite sound acquired by the microphone via wireless communication.

4. The sound field correction apparatus as claimed in claim 1, wherein,

when the plurality of speakers includes an internal speaker mounted in a predetermined cabinet and an external speaker disposed outside the cabinet, the correction unit corrects the output time of at least one of the internal speaker and the external speaker based on an external time difference, which is a time difference between the arrival time of the test sound output from the internal speaker and the arrival time of the test sound output from the external speaker.

5. A sound field correction method, comprising:

a step of simultaneously outputting test sounds having different frequencies for each of the plurality of speakers, respectively;

recording a sound signal including a composite sound of a plurality of test sounds collected by a microphone disposed at a predetermined position;

calculating a time difference in arrival time of the test sound from the speaker to the microphone between the plurality of speakers based on analysis results of the plurality of frequency components included in the sound signal; and

and correcting the output timing of the sound output from each speaker based on the time difference.

6. A non-volatile storage medium storing computer instructions that perform the following processes on a computer:

simultaneously outputting test sounds having different frequencies for each of the plurality of speakers, respectively;

recording a sound signal including a composite sound of a plurality of test sounds collected by a microphone disposed at a predetermined position;

calculating a time difference in arrival time of the test sound from the speaker to the microphone between a plurality of speakers based on analysis results of a plurality of frequency components contained in the sound signal; and

the output timing of the sound output from each of the speakers is corrected based on the time difference.