Sound field correction device, sound field correction method, and program

Abstract

To effectively suppress an influence due to an arrival time difference of a sound in a plurality of speakers.SOLUTION: A sound field compensation device of an embodiment, comprises: a test sound output part that outputs a test sound having a different frequency in each speaker from a plurality of speakers, respectively at the same time; a recording part that records an acoustic signal of a composite sound containing a plurality of test sounds collected by a microphone arranged at a predetermined position; a calculation part that calculates a time difference of an arrival time from the speaker of the test sound between the plurality of speakers to the microphone on the basis of an analysis result of a plurality of frequency components contained in the acoustic signal; and a correction part that corrects an output timing of a sound output from each speaker on the basis of the time difference.SELECTED DRAWING: Figure 2

Description

[Technical Field]

[0001] An embodiment of the present invention relates to a sound field correction device, a sound field correction method, and a program. [Background technology]

[0002] In an audio device that outputs sound from multiple speakers, the time it takes for the sound to reach the listening position from each speaker may differ. One method for solving this problem is to use a microphone to pick up a test sound output from the speaker, calculate the distance between the speaker and the microphone based on the microphone output signal, and adjust (delay) the output timing of the sound according to this distance. [Prior art documents] [Patent documents]

[0003] [Patent Document 1] Japanese Patent Application Laid-Open No. 2007-110357 Summary of the Invention [Problem to be solved by the invention]

[0004] The difference in sound arrival time described above occurs due to differences in distance between each speaker and the listening position, differences in communication speed due to the intervention of wireless communication, etc., but conventional technology may not be able to fully cancel out the difference in sound arrival time.

[0005] An object of the present invention is to provide a sound field correction device, a sound field correction method, and a program that can effectively suppress the influence of differences in sound arrival times between multiple speakers. [Means for solving the problem]

[0006] The sound field correction device of the embodiment includes a test sound output unit that simultaneously outputs test sounds from multiple speakers, each with a different frequency for each speaker; a recording unit that records an acoustic signal of a complex sound including the multiple test sounds picked up by a microphone placed at a predetermined position; a calculation unit that calculates the time difference in the arrival time of the test sounds from the multiple speakers to the microphone based on the analysis results of the multiple frequency components included in the acoustic signal; and a correction unit that corrects the output timing of the sound output from each speaker based on the time difference. [Brief explanation of the drawings]

[0007] [Figure 1] FIG. 1 is a block diagram illustrating an example of a hardware configuration of an audio device according to the first embodiment. [Figure 2] FIG. 2 is a block diagram showing an example of the functional configuration of the sound field correction device of the first embodiment. [Figure 3] FIG. 3 is a diagram illustrating an example of the positional relationship between the first speaker, the second speaker, and the microphone according to the first embodiment. [Figure 4] FIG. 4 is a diagram showing an example of an acoustic signal of a complex sound according to the first embodiment. [Figure 5] FIG. 5 is a diagram illustrating an example of a result of FFT processing on an acoustic signal according to the first embodiment. [Figure 6] FIG. 6 is a diagram showing an example of the extraction result of the first frequency and the extraction result of the second frequency according to the first embodiment. [Figure 7] FIG. 7 is a flowchart showing an example of processing in the sound field correction device according to the first embodiment. [Figure 8] FIG. 8 is a block diagram illustrating an example of a hardware configuration of the audio device according to the second embodiment. [Figure 9] FIG. 9 is a block diagram illustrating an example of the functional configuration of the sound field correction device according to the second embodiment. [Figure 10] FIG. 10 is a flowchart showing a first example of the processing in the sound field correction device according to the second embodiment. [Figure 11]FIG. 11 is a flowchart showing a second example of the processing in the sound field correction device according to the second embodiment. DETAILED DESCRIPTION OF THE INVENTION

[0008] Illustrative embodiments of the present invention are disclosed below.

[0009] (First embodiment) 1 is a block diagram showing an example of the hardware configuration of an audio device 1 according to the first embodiment. The audio device 1 is a device capable of outputting sound from multiple speakers, and may be, for example, a stereo, a video playback device, a recording device, a television, a home theater system, etc.

[0010] The acoustic device 1 according to this embodiment includes a sound field correction device 5, a first speaker 31A, a second speaker 31B, and a remote control 41. The sound field correction device 5 executes a sound field correction process that optimizes the output timing of the sounds AS1 and AS2 output from the first speaker 31A and the second speaker 31B, respectively, depending on the positional relationship between the first speaker 31A, the second speaker 31B, and the user's listening position. The listening position in this embodiment is assumed to be the position where the remote control 41 is located.

[0011] The sound field correction device 5 of this embodiment includes a CPU (Central Processing unit) 11, a memory 12, a storage 13, a user I / F (Interface) 14, a communication I / F 15, an audio decoder 21, an audio input ADC (Analog to Digital Converter) 22, a DSP (Digital Signal Processor) 23, a first delay circuit 25A, and a second delay circuit 25B, and these components are connected to each other so that they can communicate with each other via a communication bus 20.

[0012] The CPU 11 executes predetermined arithmetic and control processes in accordance with programs (including firmware, application software, etc.) stored in the memory 12, etc. The memory 12 is a main storage device including RAM (Random Access Memory), ROM (Read Only Memory), etc., and functions as a program storage area, a work area for the CPU 11, etc. The storage 13 is an auxiliary storage device including non-volatile memory such as an SSD (Solid State Drive) or HDD (Hard Disk Drive), and enables writing and reading of various data. The user I / F 14 is a device that enables receiving input from a user and outputting information to the user, and may be, for example, a display, input buttons, etc. The communication I / F 15 is a device that enables communication with other electronic devices connected via a predetermined communication network. In this embodiment, the communication I / F 15 establishes wireless communication with the remote control 41 in accordance with a predetermined standard.

[0013] The audio decoder 21 is a device that converts audio data recorded on a predetermined medium (e.g., a CD, a DVD, a Blu-ray (registered trademark) disc, a removable medium, etc.), audio data included in broadcast waves, audio data acquired from a network such as a CSP (Communications Service Provider), etc., into a digital signal in a format that can be output from the first speaker 31A and the second speaker 31B. The audio input ADC 22 is a device that converts an analog audio signal input from an external device into a digital signal.

[0014] The DSP 23 is a processor that performs predetermined processing on digital signals corresponding to sounds output from the first speaker 31A and the second speaker 31B, and generates audio signals for sounds AS1 and AS2 to be listened to, and audio signals for first test sound TS1 and second test sound TS2 (described later), etc. The first delay circuit 25A is a circuit that delays the output timing of sound AS1 output from the first speaker 31A (e.g., the Lch speaker) in accordance with a correction signal (delay signal) output from the CPU 11. The second delay circuit 25B is a circuit that delays the output timing of sound AS2 output from the second speaker 31B (e.g., the Rch speaker) in accordance with a correction signal output from the CPU 11.

[0015] The remote control 41 is a device that can be operated by a user who wants to listen to the audio 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 picked-up 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 format that enables wireless communication in accordance with 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 device 5. In addition to the above, the remote control 41 is provided with buttons and the like that accept operations by the user, but a description of these will be omitted here. Furthermore, although the present embodiment illustrates a configuration in which the microphone 45 is provided in the remote control 41, the microphone 45 may be an independent device.

[0016] During sound field correction processing, the sound field correction device 5 of this embodiment simultaneously outputs a first test sound TS1 and a second test sound TS2 from the first speaker 31A and the second speaker 31B, respectively. The first test sound TS1 and the second test sound TS2 have different frequencies. A microphone 45 mounted on the remote control 41 picks up a complex sound including the first test sound TS1 and the second test sound TS2, and a transmitter 47 transmits an acoustic signal St of the complex sound picked up by the microphone 45 to the sound field correction device 5. Based on the analysis results of the frequency components included in the acoustic signal St received from the remote control 41, the sound field correction device 5 calculates the time difference between the arrival time of the first test sound TS1 from the first speaker 31A to the microphone 45 and the arrival time of the second test sound TS2 from the second speaker 31B to the microphone 45. Then, based on the calculated time difference, the sound field correction device 5 executes a process to correct at least one of the output timing of the audio AS1 from the first speaker 31A and the output timing of the audio AS2 from the second speaker 31B, i.e., a process to control the first delay circuit 25A or the second delay circuit 25B.

[0017] Fig. 2 is a block diagram showing an example of the functional configuration of the sound field correction device 5 of the first embodiment. The sound field correction device 5 according to this embodiment has an audio output unit 101, a test sound output unit 102, a receiving unit 103, a recording unit 104, a calculation unit 105, and a correction unit 106. These functional components 101 to 106 can be realized by cooperation between hardware and software (programs) as exemplified in Fig. 1. Furthermore, at least some of these functional components 101 to 106 may be realized by dedicated hardware (circuits, etc.).

[0018] The audio output unit 101 outputs audio AS1 and AS2 to be listened to from the first speaker 31A and the second speaker 31B, respectively.

[0019] During the execution of the sound field correction process, the test sound output unit 102 outputs a first test sound TS1 from the first speaker 31A and a second test sound TS2 from the second speaker 31B. The first test sound TS1 and the second test sound TS2 have different frequencies, and the first test sound TS1 and the second test sound TS2 are output simultaneously. The test sound output unit 102 may also repeatedly output a test sound group including the first test sound TS1 and the second test sound TS2 that are output simultaneously multiple times at predetermined time intervals. For example, during the execution period of the sound field correction process, a test sound group including a set of the first test sound TS1 and the second test sound TS2 that are output simultaneously may be output multiple times (e.g., five times) at predetermined time intervals. This increases the amount of information used in the frequency analysis process, which will be described later, and improves the accuracy of the sound field correction process.

[0020] The receiving unit 103 receives an acoustic signal St of a composite sound including a first test sound TS1 and a second test sound TS2. The receiving unit 103 of this embodiment receives the acoustic signal St via wireless communication from the remote control 41. Note that the receiving unit 103 may also receive the acoustic signal St via wired communication.

[0021] The recording unit 104 records the acoustic signal St received by the receiving unit 103 in a predetermined storage device (for example, the storage 13, etc.).

[0022] The calculation unit 105 performs frequency analysis processing on the audio signal St recorded by the recording unit 104, analyzing multiple frequency components contained in the audio signal St. Based on the analysis results of the frequency analysis processing, the calculation unit 105 then calculates the time difference ΔT between the arrival time of the first test sound TS1 from the first speaker 31A to the microphone 45 and the arrival time of the second test sound TS2 from the second speaker 31B to the microphone 45. Furthermore, when a test sound group is repeatedly output multiple times, the calculation unit 105 performs the frequency analysis processing for each test sound group. This improves the accuracy of the time difference ΔT.

[0023] The correction unit 106 corrects the output timings of the voices AS1 and AS2 output from the first speaker 31A and the second speaker 31B, respectively, based on the time difference ΔT calculated by the calculation unit 105. For example, when the arrival time of the first test sound TS1 is ΔT later than the arrival time of the second test sound TS2, the output timing of the voice AS2 is delayed by ΔT from the output timing of the voice AS1. Further, when the test sound group is output repeatedly a plurality of times, the correction unit 106 corrects the output timing based on a plurality of time differences ΔT (for example, the average value of a plurality of time differences Δ) calculated for each test sound group. Thereby, the correction accuracy of the output timing can be improved.

[0024] With the above configuration, the arrival time difference between each of the voices AS1 and AS2 due to the positional relationship between the plurality of speakers (the first speaker 31A and the second speaker 31B in the present embodiment) and the listening position, communication delay due to the intervention of wireless communication, etc. can be corrected with high accuracy.

[0025] FIG. 3 is a diagram showing an example of the positional relationship among the first speaker 31A, the second speaker 31B, and the microphone 45 according to the first embodiment. In FIG. 3, a case where the distance D1 between the first speaker 31A and the microphone 45 is larger than the distance D2 between the second speaker 31B and the microphone 45 is illustrated. Also, it is shown that the first frequency Ft1 of the first test sound TS1 output from the first speaker 31A is different from the second frequency Ft2 of the second test sound TS2 output from the second speaker 31B. Here, a case where Ft1 < Ft2 is illustrated, but Ft1 > Ft2 may also be possible.

[0026] In the above situation, when the first test sound TS1 and the second test sound TS2 are output simultaneously from the first speaker 31A and the second speaker 31B, respectively, the second test sound TS2 arrives at the microphone 45 earlier than the first test sound TS1. As a result, the acoustic signal St of the complex sound picked up by the microphone 45 contains a time period that includes a component of the second frequency Ft2 but does not include a component of the first frequency Ft1, and a time period that includes both components of the first frequency Ft1 and the second frequency Ft2. The time difference ΔT can be calculated by analyzing the frequency components included in such an acoustic signal St using an appropriate method such as FFT (Fast Fourier Transform).

[0027] Fig. 4 is a diagram showing an example of an acoustic signal St of a complex sound according to the first embodiment. In the graph shown in Fig. 4, the horizontal axis represents the elapsed time from a predetermined reference time (e.g., the time when the first test sound TS1 and the second test sound TS2 are output, or the time when recording of the complex sound begins), and the vertical axis represents the signal intensity of the complex sound picked up by the microphone 45. In Fig. 4, arrival time T1 corresponds to the time it takes for the first test sound TS1 to reach the microphone 45 from the first speaker 31A, and arrival time T2 corresponds to the time it takes for the second test sound TS2 to reach the microphone 45 from the second speaker 31B. The time difference ΔT is the difference between arrival time T1 and arrival time T2.

[0028] The time difference ΔT can be calculated using any suitable method. An example of calculating the time difference ΔT using FFT is shown below.

[0029] Fig. 5 is a diagram showing an example of the result of FFT processing on the acoustic signal St according to the first embodiment, which illustrates an analysis result F when FFT processing is performed on a time window W of a predetermined time frame set on the acoustic signal St.

[0030] In the analysis result F, the horizontal axis corresponds to frequency and the vertical axis corresponds to the power spectrum. In this embodiment, a power value corresponding to a predetermined target frequency range R is calculated from the analysis result F. The target frequency range R is set based on the first frequency Ft1 when extracting the first frequency Ft1, and based on the second frequency Ft2 when detecting the second frequency Ft2. For example, the target frequency range R when extracting the first frequency Ft1 is set as (Ft1±x%). For example, when Ft1=500 (Hz) and x=10, R=500±50 (Hz). Similarly, the target frequency range R when extracting the second frequency Ft2 is set as (Ft2±x%). For example, when Ft2=2000 (Hz) and x=10, R=2000±200 (Hz). By the above processing, for each of a plurality of time windows W that are gradually shifted at predetermined time intervals, a power value corresponding to the first frequency Ft1 (e.g., R=500±50) and a power value corresponding to the second frequency Ft2 (R=2000±200) are obtained.

[0031] FIG. 6 shows an example of an extraction result F1 of the first frequency Ft1 and an extraction result F2 of the second frequency Ft2 according to the first embodiment. In the extraction result F1, the horizontal axis corresponds to time, and the vertical axis corresponds to the power value corresponding to the target frequency range R (e.g., 500±50) of the first frequency Ft1. In the extraction result F2, the horizontal axis corresponds to time, and the vertical axis corresponds to the power value corresponding to the target frequency range R (e.g., 2000±200) of the second frequency Ft2. The time axis of the extraction result F1 and the time axis of the extraction result F2 coincide with each other. In the extraction result F1, the time when the power value reaches the threshold value Th1 corresponds to the arrival time T1 of the first test sound TS1 from the first speaker 31A to the microphone 45. In the extraction result F2, the time when the power value reaches the threshold value Th2 corresponds to the arrival time T2 of the second test sound TS2 from the second speaker 31B to the microphone 45. The difference between the arrival time T1 and the arrival time T2 thus determined is the time difference ΔT.

[0032] 7 is a flowchart showing an example of processing in the sound field correction device 5 according to the first embodiment. When the sound field correction processing is started, the test sound output unit 102 outputs a first test sound TS1 from the first speaker 31A and outputs a second test sound TS2 from the second speaker 31B (S101). At this time, the first test sound TS1 and the second test sound TS2 are output simultaneously. A composite sound including the first test sound TS1 and the second test sound TS2 is picked up by the microphone 45 arranged at the listening position, and an acoustic signal St of the composite sound is recorded by the recording unit 104 (S102).

[0033] The calculation unit 105 then extracts a first frequency component corresponding to the first test sound TS1 and a second frequency component corresponding to the second test sound TS2 from the acoustic signal St using a technique such as FFT (S103). The calculation unit 105 calculates an arrival time T1 for the first speaker 31A (the time it takes for the first test sound TS1 to reach the microphone 45 from the first speaker 31A) and an arrival time T2 for the second speaker 31B (the time it takes for the second test sound TS2 to reach the microphone 45 from the second speaker 31B) (S104). The calculation unit 105 calculates a time difference ΔT from the difference between the arrival times T1 and T2 (S105). The correction unit 106 then controls the first delay circuit 25A and the second delay circuit 25B based on the time difference ΔT so that the output timing of the sound AS1 from the first speaker 31A and the output timing of the sound AS2 from the second speaker 31B are optimized (so that the time difference ΔT is reduced) (S106).

[0034] According to the above embodiment, the time difference ΔT is calculated based on the analysis results of the frequency components contained in the recorded complex sound, i.e., based on the comparison results of the times at which each frequency component is detected in the recorded acoustic signal St. This makes it possible to eliminate the effects of the time from when a command to output a test sound is issued until the test sound is actually output from the speaker, or the time from when the test sound is picked up by a microphone until it is recorded, etc. This effectively reduces the effects of the difference in the time it takes for sounds from multiple speakers to reach the listening position, making it possible to provide a comfortable sound field.

[0035] Other embodiments will be described below, but descriptions of parts that have the same or similar effects as the first embodiment will be omitted as appropriate.

[0036] (Second embodiment) 8 is a block diagram showing an example of the hardware configuration of an audio device 2 according to the second embodiment. In this embodiment, the audio device 2 includes a first internal speaker 38A and a second internal speaker 38B, and an external speaker 51 is connected to the audio device 2. The first internal speaker 38A and the second internal speaker 38B are, for example, speakers mounted in a predetermined housing (such as a housing constituting the main body of the audio device 2). The external speaker 51 is, for example, a speaker disposed outside the housing in which the first internal speaker 38A and the second internal speaker 38B are mounted.

[0037] The sound field correction device 6 of this embodiment simultaneously outputs a first test sound TS1, a second test sound TS2, and a third test sound TS3 from the first internal speaker 38A, the second internal speaker 38B, and the external speaker 51, respectively. The sound field correction device 5 executes a sound field correction process that optimizes the output timing of sounds AS1, AS2, and AS3 output from the first internal speaker 38A, the second internal speaker 38B, and the external speaker 51, respectively, depending on the arrival time difference between the first test sound TS1, the second test sound TS2, and the third test sound TS3 until they reach the microphone 45.

[0038] The audio device 2 of this embodiment includes a transmitter 27 that establishes wireless communication with the external speaker 51. The sound field correction device 6 of this embodiment includes a third delay circuit 25C that is connected to the transmitter 27. The output timing of the audio AS3 from the external speaker 51 can be adjusted by the action of the third delay circuit 25C. Note that, although this embodiment illustrates a configuration in which the audio device 2 and the external speaker 51 are connected via wireless communication, the audio device 2 and the external speaker 51 may also be connected via wired communication.

[0039] 9 is a block diagram showing an example of the functional configuration of a sound field correction device 6 according to the second embodiment. The audio output unit 101 of this embodiment outputs sounds AS1, AS2, and AS3 to be listened to from the first internal speaker 38A, the second internal speaker 38B, and the external speaker 51, respectively. Furthermore, the audio output unit 101 of this embodiment simultaneously outputs the first test sound TS1, the second test sound TS2, and the third test sound TS3 from the first internal speaker 38A, the second internal speaker 38B, and the external speaker 51, respectively, during execution of the sound field correction process.

[0040] The recording unit 104 of this embodiment records an audio signal St of a composite sound including at least two of the first test sound TS1, the second test sound TS2, and the third test sound TS3. The calculation unit 105 of this embodiment calculates an internal time difference ΔTint, which is the time difference between the arrival time T1 of the first test sound TS1 and the arrival time T2 of the second test sound TS2, and an external time difference ΔText, which is the time difference between the arrival time (e.g., arrival time T1) of at least one of the first test sound TS1 and the second test sound TS2 and the arrival time T3 of the third test sound TS3, based on the results of frequency analysis of the audio signal St. The correction unit 106 of this embodiment corrects the output timing of at least one of the first internal speaker 38A and the second internal speaker 38B based on the internal time difference ΔTint, as in the first embodiment. Furthermore, correction unit 106 of this embodiment corrects the output timing of at least one of first internal speaker 38A, second internal speaker 38B, and external speaker 51 based on external time difference ΔText.

[0041] 10 is a flowchart showing a first example of the processing in the sound field correction device 6 according to the second embodiment. In this example, the sound field correction processing is performed using two types of test sounds.

[0042] When the sound field correction process starts, the test sound output unit 102 outputs the first test sound TS1 from the first internal speaker 38A and outputs the second test sound TS2 from the second internal speaker 38B (S201). At this time, the first test sound TS1 and the second test sound TS2 are output simultaneously. A composite sound including the first test sound TS1 and the second test sound TS2 is picked up by the microphone 45 arranged at the listening position, and an acoustic signal St of the composite sound is recorded by the recording unit 104 (S202).

[0043] The calculation unit 105 then extracts a first frequency component corresponding to the first test sound TS1 and a second frequency component corresponding to the second test sound TS2 from the acoustic signal St using a technique such as FFT (S203). The calculation unit 105 calculates the arrival time T1 of the first internal speaker 38A (the time it takes for the first test sound TS1 to travel from the first internal speaker 38A to the microphone 45) and the arrival time T2 of the second internal speaker 38B (the time it takes for the second test sound TS2 to travel from the second internal speaker 38B to the microphone 45) (S204). The calculation unit 105 calculates the internal time difference ΔTint from the difference between the arrival time T1 and the arrival time T2 (S205).

[0044] Thereafter, the test sound output unit 102 outputs the first test sound TS1 from the first internal speaker 38A, and outputs the second test sound TS2 as the third test sound TS3 from the external speaker 51 (S206). A composite sound including the first test sound TS1 and the second test sound TS2 is picked up by the microphone 45 arranged at the listening position, and an acoustic signal St of the composite sound is recorded by the recording unit 104 in the storage 13 or the like (S207).

[0045] The calculation unit 105 then extracts a first frequency component corresponding to the first test sound TS1 and a second frequency component corresponding to the second test sound TS2 from the acoustic signal St using a technique such as FFT (S208). The calculation unit 105 calculates an arrival time T1 of the first internal speaker 38A (the time it takes for the first test sound TS1 to travel from the first internal speaker 38A to the microphone 45) and an arrival time T3 of the external speaker 51 (the time it takes for the second test sound TS2 to travel from the external speaker 51 to the microphone 45) (S209). The calculation unit 105 calculates an external time difference ΔText from the difference between the arrival time T1 and the arrival time T3 (S210).

[0046] Then, based on the internal time difference ΔTint and the external time difference ΔText, the correction unit 106 controls the first delay circuit 25A, the second delay circuit 25B, and the third delay circuit 25C so that the output timing of audio AS1 from the first internal speaker 38A, the output timing of audio AS2 from the second internal speaker 38B, and the output timing of audio AS3 from the external speaker 51 are optimized (S211).

[0047] According to the above process, two types of test sounds (first test sound TS1 and second test sound TS2) are used to perform sound field correction processing on a sound field including three speakers (first internal speaker 38A, second internal speaker 38B, and external speaker 51). This allows frequency analysis processing to be performed using a relatively simple algorithm.

[0048] 11 is a flowchart showing a second example of the processing in the sound field correction device 6 according to the second embodiment. In this example, the sound field correction processing is performed using three types of test sounds.

[0049] When the sound field correction process is started, the test sound output unit 102 outputs the first test sound TS1 from the first internal speaker 38A, the second test sound TS2 from the second internal speaker 38B, and the third test sound TS3 from the external speaker 51 (S301). At this time, the first test sound TS1, the second test sound TS2, and the third test sound TS3 are output simultaneously. A composite sound including the first test sound TS1, the second test sound TS2, and the third test sound TS3 is picked up by the microphone 45 arranged at the listening position, and an acoustic signal St of the composite sound is recorded by the recording unit 104 (S302).

[0050] The calculation unit 105 then extracts from the acoustic signal St a first frequency component corresponding to the first test sound TS1, a second frequency component corresponding to the second test sound TS2, and a third frequency component corresponding to the third test sound TS3 using a technique such as FFT (S303). The calculation unit 105 calculates the arrival time T1 of the first internal speaker 38A (the time it takes for the first test sound TS1 to reach the microphone 45 from the first internal speaker 38A), the arrival time T2 of the second internal speaker 38B (the time it takes for the second test sound TS2 to reach the microphone 45 from the second internal speaker 38B), and the arrival time T3 of the external speaker 51 (the time it takes for the third test sound TS3 to reach the microphone 45 from the external speaker 51) (S304). The calculation unit 105 calculates the internal time difference ΔTint from the difference between the arrival time T1 and the arrival time T2, and calculates the 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.

[0051] Then, based on the internal time difference ΔTint and the external time difference ΔText, the correction unit 106 controls the first delay circuit 25A, the second delay circuit 25B, and the third delay circuit 25C so that the output timing of audio AS1 from the first internal speaker 38A, the output timing of audio AS2 from the second internal speaker 38B, and the output timing of audio AS3 from the external speaker 51 are optimized (S306).

[0052] According to the above process, three types of test sounds (first test sound TS1, second test sound TS2, and third test sound TS3) are output simultaneously, and sound field correction processing can be performed on a sound field including three speakers (first internal speaker 38A, second internal speaker 38B, and external speaker 51). This allows the sound field correction processing to be performed quickly.

[0053] The programs for realizing the functions of the sound field correction devices 5 and 6 described above may be provided by being recorded in a computer-readable recording medium such as a CD-ROM, a flexible disk (FD), a CD-R, or a digital versatile disk (DVD) in the form of a computer-installable or executable file. Alternatively, the programs may be stored on a computer connected to a network such as the Internet and provided by being downloaded via the network. Alternatively, the programs may be provided or distributed via a network such as the Internet.

[0054] Although the embodiments of the present invention have been described above, these embodiments are presented as examples and are not intended to limit the scope of the invention. These novel embodiments can be embodied in various other forms, and various omissions, substitutions, and modifications can be made without departing from the spirit of the invention. These embodiments and their modifications are included within the scope and spirit of the invention, and are also included in the inventions and their equivalents as defined in the claims. [Explanation of symbols]

[0055] 1,2...acoustic device, 5,6...sound field correction device, 11...CPU, 12...memory, 13...storage, 14...user I / F, 15...communication I / F, 21...audio decoder, 22...audio input ADC, 23...DSP, 25A...first delay circuit, 25B...second delay circuit, 25C...third delay circuit, 31A...first speaker, 31B...second speaker, 38A...first internal speaker, 38B...second internal speaker, 41...remote control, 45...microphone, 46...wireless Modulation circuit, 47...transmitter, 101...audio output unit, 102...test sound output unit, 103...receiving unit, 104...recording unit, 105...calculating unit, 106...correcting unit, AS1, AS2...audio, F...analysis result, F1, F2...extraction result, R...target frequency range, St...acoustic signal, T1, T2...arrival time, TS1...first test sound, TS2...second test sound, TS3...third test sound, W...time window, ΔT...time difference, ΔTint...internal time difference, ΔText...external time difference

Claims

1. a test sound output unit that simultaneously outputs test sounds having different frequencies from each of a plurality of speakers; a recording unit that records an acoustic signal of a complex sound including the plurality of test sounds picked up by a microphone disposed at a predetermined position; a calculation unit that calculates a time difference between arrival times of the test sound from the speakers to the microphone among the plurality of speakers based on an analysis result of a plurality of frequency components included in the acoustic signal; a correction unit that corrects the output timing of the sound output from each of the speakers based on the time difference; A sound field correction device comprising:

2. the test sound output unit repeatedly outputs a test sound group including a plurality of the test sounds to be output simultaneously at predetermined time intervals; the correction unit corrects the output timing based on the plurality of time differences calculated for each of the test sound groups. The sound field correction device according to claim 1 .

3. a receiving unit that receives the acoustic signal of the complex sound picked up by the microphone via wireless communication; The sound field correction device according to claim 1 , further comprising:

4. When the plurality of speakers include an internal speaker mounted in a predetermined housing and an external speaker disposed outside the housing, the correction unit corrects the output timing 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. The sound field correction device according to claim 1 .

5. a step of simultaneously outputting test sounds from a plurality of speakers, each test sound having a different frequency for each speaker; a step of recording an acoustic signal of a complex sound including a plurality of the test sounds picked up by a microphone arranged at a predetermined position; calculating a time difference between arrival times of the test sound from the plurality of speakers to the microphone based on an analysis result of a plurality of frequency components included in the acoustic signal; correcting the output timing of the sound output from each of the speakers based on the time difference; A sound field correction method including:

6. On the computer, a process of simultaneously outputting test sounds from a plurality of speakers, each test sound having a different frequency; a process of recording an acoustic signal of a complex sound including a plurality of the test sounds picked up by a microphone arranged at a predetermined position; a process of calculating a time difference between arrival times of the test sound from the plurality of speakers to the microphone based on an analysis result of a plurality of frequency components included in the acoustic signal; a process of correcting the output timing of the sound output from each of the speakers based on the time difference; A program that executes the following.

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