JP5938771B2 - Test method and test apparatus - Google Patents

Description

  The present case relates to a test method and a test apparatus.

  A technique for testing a test target device equipped with a plurality of sound output devices (speakers) is disclosed. For example, Patent Documents 1 and 2 disclose a technique for determining a speaker connection state of an in-vehicle audio system including a plurality of speakers.

JP-A-5-49096 JP-A-3-283800

  However, with the techniques of Patent Documents 1 and 2, it is difficult to perform a sound test when a test target device equipped with a plurality of sound output devices is conveyed on a conveyor line or the like.

  The present invention has been made in view of the above problems, and an object thereof is to provide a test method and a test apparatus capable of performing a sound test when a test target device equipped with a plurality of sound output devices is transported.

  In the test method disclosed in the specification, a specific frequency component is detected by using a sound sensor that moves relative to a test target device on which a plurality of sound output devices are mounted. The sound source signal output to the signal is separated and extracted.

  The test apparatus disclosed in the specification includes a sound sensor that moves relative to a test target device including a plurality of sound output devices, and a specific frequency component detected by a signal detected by the sound sensor, An extraction unit that separates and extracts sound source signals output in order by the plurality of sound output devices.

  According to the test method and the test apparatus disclosed in the specification, a sound test can be performed when a device on which a plurality of sound output devices are mounted is transported.

It is a figure for demonstrating the outline of the testing apparatus used for a sound test. (A) is a block diagram for demonstrating the structure of the test apparatus and test object apparatus which concern on Example 1, (b) is a block diagram showing each function of a calculating part. It is a schematic diagram for demonstrating the positional relationship of a recording start sensor, a microphone, and a test object apparatus. It is a block diagram for demonstrating the hardware constitutions of a calculating part. (A) is a figure showing the waveform of the chirp signal changed continuously from low frequency to high frequency, (b) is a figure showing the waveform of the chirp signal output from a speaker. (A) is a first channel sound source signal for driving the first speaker, and (b) is a second channel sound source signal for driving the second speaker. (A) is an input signal for one period input to the microphone, and (b) is an extraction component of the specific frequency (Fp) extracted by the channel extraction unit based on the result of the Fourier transform by the Fourier transform unit. . It is a figure showing the time change of the extraction component of a specific frequency (Fp). It is a figure showing the time change of the extraction component of a specific frequency (Fp) when a 1st channel sound source signal is reproduced | regenerated by a 2nd speaker and a 2nd channel sound source signal is reproduced | regenerated by a 1st speaker. (A) is an example of a chirp signal to be extracted by the waveform extraction unit, and (b) is a speaker frequency characteristic graph obtained by the Fourier transform unit for the chirp signal to be extracted by the waveform extraction unit. (A) is a figure showing the relationship between a chirp signal and its specific frequency extraction component, (b) is a figure showing the relationship between the amplitude change of a chirp signal accompanying a movement, and the change of each peak amplitude value of a specific frequency extraction component. is there. (A) is a figure for demonstrating an example of the approximated curve calculated from each peak amplitude value, (b) is a figure showing a recording signal and a volume correction curve, (c) is a recording signal after correction | amendment. FIG. It is a figure for demonstrating an example of the flowchart performed in the case of the sound test by a test device. (A) And (b) is an example of sound source signal data when sound quality characteristic evaluation is not performed. It is a figure for demonstrating the example of the flowchart performed when performing a sound test using the sound source signal of Fig.14 (a). (A) is a block diagram for demonstrating the structure of a test object apparatus, (b) is a schematic diagram for demonstrating the positional relationship of a recording start sensor, a microphone, and a test object apparatus. It is a figure for demonstrating the other example of the flowchart performed when performing a sound test using the sound source signal of Fig.14 (a).

  Prior to the description of the examples, the sound test will be described. The sound test is a test in which a personal computer, a mobile phone, an audio device, or the like is used as a test target device and the sound output of a sound output device provided in the test target device is inspected. In this case, a device equipped with a plurality of sound output devices is a test target device. The plurality of sound output devices may be provided so as to play in a one-to-one correspondence with the plurality of channel sound sources. For example, left and right speakers provided corresponding to left and right channel sound sources may be used.

  Fig.1 (a) is a figure for demonstrating the outline of the test apparatus used for a sound test. A case where a plurality of speakers 2a and 2b are provided in a personal computer will be described with reference to FIG. For example, it is assumed that the speaker 2a reproduces the sound source of the first channel and the speaker 2b reproduces the sound source of the second channel. In the sound test, the sound signal output from the speakers 2a and 2b on the conveyor line 1 is detected by the sound sensor (microphone) 3, and the detected sound signal is measured.

  In order to accurately measure the sound signals output from the speakers 2a and 2b, it is preferable that the speakers 2a and 2b and the microphone 3 are close to each other. Therefore, a plurality of fixed position sensors 4a and 4b may be provided. For example, referring to FIG. 1B, the position sensor 4a detects a case where the distance between the microphone 3 and the speaker 2a falls within a predetermined range by detecting passage of a predetermined portion of the personal computer. By reproducing only the speaker 2a at the position, the sound signal of the speaker 2a can be accurately measured. Referring to FIG. 1C, the position sensor 4b detects when the distance between the microphone 3 and the speaker 2b falls within a predetermined range by detecting passage of a predetermined location of the personal computer. By reproducing only the speaker 2b at the position, the sound signal of the speaker 2b can be accurately measured.

  However, in the test apparatus shown in FIGS. 1B and 1C, the signal line of the sensor output is directly connected between the test object apparatus moving on the conveyor line 1 and the fixed position sensors 4a and 4b. I can't connect. Therefore, notification from the position sensors 4a and 4b to the target device requires notification by wireless communication, a keyboard key press, and the like, and the communication equipment is expensive. In addition, when the test target devices cover a variety of models having different shapes, it is necessary to change the fixed positions of the position sensors 4a and 4b for each model. Also, it is necessary to synchronize the start of playback with the start of recording. However, accurate synchronization is difficult due to delays in wireless communication, keystroke machine operation, and software processing.

  Therefore, in the following embodiments, a test method and a test apparatus capable of performing a sound test when a test target device equipped with a plurality of sound output devices is transported will be described.

  FIG. 2A is a block diagram for explaining the configuration of the test apparatus 100 and the test target device 200 according to the first embodiment. Referring to FIG. 2A, the test apparatus 100 includes a recording start sensor 10, a microphone 20, an input amplifier 30, an A / D conversion unit 40, and a calculation unit 50. The test target device 200 includes a sound source data storage unit 201, a control unit 202, a first D / A converter 203a, a second D / A converter 203b, a first output amplifier 204a, a second output amplifier 204b, a first speaker 205a, and A second speaker 205b is provided.

  FIG. 2B is a block diagram illustrating each function of the calculation unit 50. The calculation unit 50 includes a recording data storage unit 51, a sound source information storage unit 52, a channel extraction unit 53, a waveform extraction unit 54, a noise processing unit 55, a Fourier transform unit 56, a determination unit 57, an output unit 58, and a volume correction unit 59. Function.

  FIG. 3 is a schematic diagram for explaining the positional relationship among the recording start sensor 10, the microphone 20, and the test target device 200. With reference to FIG. 3, the test target device 200 is placed on a belt conveyor line 206. It is preferable that the test target device 200 is placed so that the shortest distance between the microphone 20 and the first speaker 205a is the same as the shortest distance between the microphone 20 and the second speaker 205b in the course of conveyance. For example, the test target device 200 is preferably arranged so that a line connecting the center of the first speaker 205 a and the center of the second speaker 205 b is parallel to the conveying direction of the belt conveyor line 206. The recording start sensor 10 and the microphone 20 are disposed above the belt conveyor line 206. The microphone 20 is disposed downstream of the recording start sensor 10 in the moving direction of the belt conveyor line 206.

  FIG. 4 is a block diagram for explaining the hardware configuration of the arithmetic unit 50. Referring to FIG. 4, the calculation unit 50 includes a CPU 101, a RAM 102, a storage device 103, an interface 104, and the like. Each of these devices is connected by a bus or the like. A CPU (Central Processing Unit) 101 is a central processing unit. The CPU 101 includes one or more cores. A RAM (Random Access Memory) 102 is a volatile memory that temporarily stores programs executed by the CPU 101, data processed by the CPU 101, and the like. The storage device 103 is a nonvolatile storage device. As the storage device 103, for example, a ROM (Read Only Memory), a solid state drive (SSD) such as a flash memory, a hard disk driven by a hard disk drive, or the like can be used. When the CPU 101 executes a predetermined program, the calculation unit 50 includes a recording data storage unit 51, a sound source information storage unit 52, a channel extraction unit 53, a waveform extraction unit 54, a noise processing unit 55, a Fourier transform unit 56, and a determination unit. 57, functions as an output unit 58, and a sound volume correction unit 59.

  Next, the operation of each part of the test target device 200 and the test apparatus 100 will be described with reference to FIGS. The sound source data storage unit 201 is a non-volatile memory such as a ROM (read only memory), a hard disk, or a flash memory, and stores sound source signal data for a sound test. The control unit 202 includes a CPU (Central Processing Unit), a RAM (Random Access Memory), and the like, and generates a sound source signal for each channel according to sound source signal data stored in the sound source data storage unit 201. In the present embodiment, the control unit 202 generates two types (two channels) of sound source signals, a first channel sound source signal and a second channel sound source signal.

  The first D / A converter 203a and the second D / A converter 203b are digital / analog converters that convert a digital signal into an analog signal such as current, and the sound source signal generated by the control unit 202 is converted into an analog signal such as current. Convert to In the present embodiment, the first D / A converter 203a converts the first channel sound source signal into an analog signal, and the second D / A converter 203b converts the second channel sound source signal into an analog signal.

  The first output amplifier 204a and the second output amplifier 204b are amplifiers. The first output amplifier 204a amplifies the analog signal generated by the first D / A converter 203a. The second output amplifier 204b amplifies the analog signal generated by the second D / A converter 203b. The first speaker 205a and the second speaker 205b output sound according to the amplified analog signal. The first speaker 205a outputs sound according to the analog signal amplified by the first output amplifier 204a. The second speaker 205b outputs sound according to the analog signal amplified by the second output amplifier 204b.

  The recording start sensor 10 is a sensor that detects the passage of an object. In the present embodiment, the recording start sensor 10 detects whether or not passage of a predetermined portion of the test target device 200 is started in the moving direction of the belt conveyor line 206. When the recording start sensor 10 detects the start of passage of a predetermined portion of the test target device 200, the recording start sensor 10 inputs a signal related to the detection to the calculation unit 50. The microphone 20 is a sound sensor and converts sound output from the first speaker 205a and the second speaker 205b into an analog signal such as a voltage. The input amplifier 30 is an amplifier, and amplifies the analog signal output from the microphone 20. The A / D converter 40 is an analog / digital converter that converts an analog signal into a digital signal, and converts the analog signal amplified by the input amplifier 30 into a digital signal.

  When the calculation unit 50 receives a signal related to detection of passage of the test target device 200 from the recording start sensor 10, the calculation unit 50 starts recording. Specifically, the recording data storage unit 51 stores a digital signal output from the A / D converter 40 as recording data. Note that the recording data storage unit 51 stops recording after a predetermined time from the start of recording. For example, the recording data storage unit 51 may stop recording at the scheduled time when the test target device 200 passes through the microphone 20 after the recording is started. The sound source information storage unit 52 stores information related to sound source signal data for sound testing. The computing unit 50 examines the sound output of the test target device 200 based on the recording data stored in the recording data storage unit 51 and the information on the sound source signal data stored in the sound source information storage unit 52. Next, details of the operation of the calculation unit 50 will be described.

  First, an example of sound source signal data for a sound test will be described. Although the sound source signal is not particularly limited, a chirp signal that can be used for a frequency characteristic test will be described as an example. The chirp signal is a swept sine wave signal having an audible range of 20 Hz to 20 kHz or the like, and is also called a swept sine wave signal. Specifically, the chirp signal is a signal whose frequency is continuously changed (swept) from a low frequency to a high frequency, or a signal whose frequency is continuously changed (swept) from a high frequency to a low frequency.

  FIG. 5A is a diagram illustrating a waveform of a chirp signal in which the frequency is continuously changed from a low frequency to a high frequency. In FIG. 5A, the horizontal axis represents time, and the vertical axis represents a drive signal (for example, current) for driving the speaker. Referring to FIG. 5A, the electric signal input to the speaker has a constant amplitude at any frequency. On the other hand, FIG.5 (b) is a figure showing the waveform of the chirp signal output from a speaker. In FIG. 5B, the horizontal axis represents time, and the vertical axis represents the volume output from the speaker. With reference to FIG.5 (b), an output becomes small with respect to the electric signal input in a low frequency area and a high frequency area according to the frequency characteristic of a speaker.

  Next, a specific example of sound source signal data will be described. The sound source signal data includes a first channel sound source signal and a second channel sound source signal. FIG. 6A shows a first channel sound source signal for driving the first speaker 205a. FIG. 6B shows a second channel sound source signal for driving the second speaker 205b. 6A and 6B, the horizontal axis represents time, and the vertical axis represents a drive signal (for example, current) for driving the speaker.

  With reference to FIG. 6A and FIG. 6B, the sound source signal data first has a cue signal. The cue signal is a single-frequency sine wave signal having a specific frequency (Fp). The specific frequency (Fp) is preferably a frequency in a region where the frequency characteristics of the first speaker 205a and the second speaker 205b are relatively stable. As the specific frequency (Fp), for example, a frequency of about 1 kHz included in the chirp signal can be used. In the example of FIG. 6A and FIG. 6B, the cue signal is output simultaneously to both the first speaker 205a and the second speaker 205b, but may be output to either one. However, it is preferable to output to both speakers so that the cue signal can be detected by the microphone 20 regardless of where the test target device 200 is located.

  The sound source signal data includes a first chirp signal to be output to the first speaker 205a after the end of the cue signal. The sound source signal data has a second chirp signal to be output to the second speaker 205b after the end of the first chirp signal. The silence time between the cue signal and the first chirp signal is defined as a silence time Tg1. The silence time between the first chirp signal and the second chirp signal is defined as a silence time Tg2. The silent time between the second chirp signal and the next signal is the silent time Tg3. The period from the start point of the cueing signal to the end point of the silence time Tg3 is defined as one period of the sound source signal data. Although one period of the sound source signal data is not particularly limited, it is sufficient that at least each channel sound source signal is included at least once in order. From the start of recording to the end of recording, the above cycle is repeated. The silent periods Tg1 to Tg3 may be zero seconds because they may be provided as necessary.

  Next, an input signal input to the microphone 20 when the sound source signal data in FIGS. 6A and 6B is output to the first speaker 205a and the second speaker 205b will be described. FIG. 7A shows an input signal for one cycle input to the microphone 20. With reference to FIG. 7A, first, a cue signal is input to the microphone 20. After the end of the cue signal, the first chirp signal is input after the gap Tg1. After the end of the first chirp signal, the second chirp signal is input after the gap Tg2. If the distance between the microphone 20 and the first speaker 205a is different from the distance between the microphone 20 and the second speaker 205b, the volume of the first chirp signal and the second chirp signal is different. In the example of FIG. 7A, the volume of the first chirp signal is smaller than the volume of the second chirp signal.

  The Fourier transform unit 56 performs frequency analysis (such as short-time Fourier transform) on the recording data stored in the recording data storage unit 51. Short-time Fourier transform (StFt: Short-time Fourier Transform) is a method of analyzing a time change of a frequency spectrum by multiplying a window function while shifting it by a minute time Δt and sequentially performing Fourier transform. The short-time Fourier transform is a method generally used for a time-varying signal such as sound. A wavelet transform may be used for the same purpose. Note that before the Fourier transform by the Fourier transform unit 56, the noise processing unit 55 may perform a noise removal process on the recorded data.

  FIG. 7B shows an extraction component of the specific frequency (Fp) extracted by the channel extraction unit 53 based on the result of the Fourier transform by the Fourier transform unit 56. Referring to FIG. 7B, since the cue signal is a signal having a specific frequency (Fp), the specific frequency (Fp) is extracted from the beginning to the end of the cue signal. Therefore, the width from the beginning to the end of the cue signal is the pulse width of the extracted component of the specific frequency (Fp). Although the first chirp signal and the second chirp signal also include the specific frequency (Fp), since the time for extracting the specific frequency (Fp) is short, the specific frequency (Fp) in the first chirp signal and the second chirp signal is extracted. The pulse width of the component is reduced.

  The specific frequency (Fp), the chirp signal duration Ts_e, the time Ts_p from the start of the chirp signal until the specific frequency (Fp) appears, and the silence times Tg1 to Tg3 are sound source information as information on sound source signal data for sound test It is stored in the storage unit 52. The duration Ts_e, the time Ts_p, and the silence times Tg1 to Tg3 can be acquired in advance.

  In the example of FIGS. 6A and 6B, the first chirp signal is reproduced after the cue signal, and the second chirp signal is reproduced after the first chirp signal. Therefore, if a narrow pulse having a specific frequency (Fp) is detected in the vicinity of Tg1 + Ts_p from the trailing edge of the wide pulse having a specific frequency (Fp), it is determined that the first chirp signal is input to the microphone 20. can do. If a narrow pulse with a specific frequency (Fp) is detected in the vicinity of Tg1 + Ts_e + Tg2 + Ts_p from the trailing edge of a wide pulse with a specific frequency (Fp), the second chirp signal is input to the microphone 20 if that pulse is detected. Judgment can be made.

  FIG. 8 is a diagram illustrating the time change of the extracted component of the specific frequency (Fp). The one-dot chain line is an envelope of the amplitude value of the extracted component of the specific frequency (Fp) of the first chirp signal. A two-dot chain line is an envelope of the amplitude value of the extracted component of the specific frequency (Fp) of the second chirp signal. The dotted line is an envelope of the amplitude value of the cue signal. Each envelope can be obtained by numerical calculation such as fitting an amplitude value to a polynomial curve.

  Referring to FIG. 8, as the first speaker 205a approaches the microphone 20, the amplitude value of the extracted component of the specific frequency (Fp) of the first chirp signal increases. When the distance between the first speaker 205a and the microphone 20 becomes the shortest, the amplitude value of the extracted component of the specific frequency (Fp) of the first chirp signal becomes the maximum value ARM. Further, the time in this case is the passage time TRM when the microphone 20 has passed through the shortest distance (for example, the center position of the first speaker 205a) from the first speaker 205a.

  Thereafter, since the microphone 20 is separated from the first speaker 205a, the amplitude value of the extracted component of the specific frequency (Fp) of the first chirp signal becomes small. On the other hand, since the microphone 20 approaches the second speaker 205b, the amplitude value of the extracted component of the specific frequency (Fp) of the second chirp signal increases. When the distance between the second speaker 205b and the microphone 20 becomes the shortest, the amplitude value of the extracted component of the specific frequency (Fp) of the second chirp signal becomes the maximum value ALM. The time in this case is the passage time TLM when the microphone 20 has passed the shortest distance of the second speaker 205b (for example, the center position of the second speaker 205b). Thereafter, since the microphone 20 is separated from the second speaker 205b, the amplitude value of the extracted component of the specific frequency (Fp) of the second chirp signal becomes small. Since the cue signal is output from both speakers, the amplitude value of the cue signal remains large over a wide range.

  By detecting the maximum value ARM, the maximum value ALM, the passage time TRM, the passage time TLM, and the like, it is possible to determine whether the test target device 200 is acceptable. For example, if the maximum value ARM and the maximum value ALM are within a predetermined pass determination range, the determination unit 57 determines the test target device 200 as a non-defective product with respect to the sound volume. In addition, it is possible to determine the pass / fail of each speaker mounting position (connection of wiring, etc.) by detecting the context of the passage time TRM and the passage time TLM.

  As an example, the case where the wiring to each speaker is switched will be described. FIG. 9 is a diagram illustrating a temporal change in the extracted component of the specific frequency (Fp) when the first channel sound source signal is reproduced by the second speaker 205b and the second channel sound source signal is reproduced by the first speaker 205a. . The one-dot chain line is an envelope of the amplitude value of the extracted component of the specific frequency (Fp) of the first chirp signal. A two-dot chain line is an envelope of the amplitude value of the extracted component of the specific frequency (Fp) of the second chirp signal. The dotted line is an envelope of the amplitude value of the cue signal.

  When the wiring to each speaker is switched, the first channel sound source signal is reproduced by the second speaker 205b, and the second channel sound source signal is reproduced by the first speaker 205a. In this case, the passage time TLM appears before the passage time TRM. Therefore, the determination part 57 determines with the wiring to each speaker having been replaced, when the passage time TLM appears before the passage time TRM.

  Next, sound quality pass / fail determination will be described. Extraction of the sound quality characteristic of the first speaker 205a can be performed by extracting the first chirp signal, and determination of the sound quality characteristic of the second speaker 205b can be performed by extracting the second chirp signal. Since the determination accuracy improves as the volume increases, it is preferable to extract the first chirp signal near the maximum value ARM, and it is preferable to extract the second chirp signal near the maximum value ALM.

  FIG. 10A shows an example of a chirp signal to be extracted by the waveform extraction unit 54. The waveform extraction unit 54 uses a recording signal from the time when the specific frequency (Fp) is detected as far as the time Ts_p as the start point, and the recording signal until the duration Ts_e elapses. Since the cue signal and the chirp signal can be distinguished according to the pulse width of the extraction component of the specific frequency (Fp), the waveform extraction unit 54 can exclude the cue signal.

  FIG. 10B is a speaker frequency characteristic graph obtained by the Fourier transform unit 56 for the chirp signal to be extracted by the waveform extraction unit 54. In FIG. 10B, the horizontal axis represents frequency and the vertical axis represents amplitude. In order to generate a sweep of the chirp signal so that the frequency of the sine wave is proportional to time or the logarithm of frequency is proportional to time, the horizontal axis of the F characteristic (frequency characteristic) graph should be replaced from time to frequency. Can do. By determining whether or not the amplitude for each frequency is within an allowable range, it is possible to determine whether the first speaker 205a and the second speaker 205b are acceptable.

  With reference to FIG.10 (b), the acceptable upper limit line and the acceptable lower limit line are drawn as the tolerance | permissible_range of the amplitude with respect to each frequency. If all the results obtained by the Fourier transform unit 56 are within the allowable range, the determination unit 57 determines that the sound quality characteristic of the target speaker is “good”. The output unit 58 outputs the determination result of the determination unit 57 to the external device.

  Since the sound volume changes as the test target device 200 moves, the sound quality characteristic may be evaluated after correcting the sound volume. Specifically, the sound volume correction unit 59 stores the necessary number of chirp signals in the recording data storage unit 51 for each channel using the duration Ts_e stored in the sound source information storage unit 52. For example, if the required number is N (N is a predetermined natural number), recording may be performed for a time equal to or longer than (N + 1) × duration Ts_e.

  Next, referring to FIG. 11A, the volume correction unit 59 reads the specific frequency (Fp) stored in the sound source information storage unit 52, and the peak amplitude value of the specific frequency (Fp) from each chirp signal. Get Ap. The sound volume correction unit 59 obtains peak amplitude values Ap1, Ap2,..., ApN at a specific frequency (Fp) for each chirp signal. Since the test target device 200 moves relative to the microphone 20, each peak amplitude value is a different value with reference to FIG.

  The sound volume correction unit 59 generates an approximate curve f (t) most applicable to the point sequence data of each peak amplitude value. The approximate curve (regression curve) can be generated by a numerical calculation method such as high-order polynomial curve fitting (Curve Fitting) such as the least square method. The approximate curve is at least a quadratic expression, and is generated from point sequence data of three (degree of polynomial + 1) or more. FIG. 12A is a diagram for explaining an example of the approximate curve calculated from each peak amplitude value.

  The maximum amplitude value Apc, which is the maximum value of the approximate curve f (t), is estimated to correspond to the sound volume when the microphone 20 passes through the point of the shortest distance from the speaker. In addition, the time Tpc corresponding to the maximum amplitude value Apc is estimated to correspond to the time when the microphone 20 has passed the shortest distance from the speaker. As described above, even when the specific frequency (Fp) does not appear at the shortest distance between the speaker and the microphone 20, it is possible to estimate the volume and time when the microphone 20 passes the point at the shortest distance from the speaker.

  The volume correction unit 59 calculates the volume correction curve [Apc / f (t)] by multiplying the inverse of the approximate curve f (t) by the maximum amplitude value Apc. Next, referring to FIG. 12B, the volume correction unit 59 corrects the recording signal by matching the time and multiplying the recording signal by the volume correction curve [Apc / f (t)]. To do. In particular, since the maximum amplitude value Apc is used, each recording signal can be corrected so that the microphone 20 and the speaker are positioned at the shortest distance. FIG. 12C shows the recording signal after correction. Referring to FIG. 12C, the influence of the relative movement of the speaker is avoided in the corrected recording signal.

  FIG. 13 is a diagram for explaining an example of a flowchart executed in the sound test by the test apparatus 100. Hereinafter, the flowchart of FIG. 13 will be described. First, the channel extraction unit 53 reads the duration Ts_e and the time Ts_p from the sound source information storage unit 52 (step S1). Next, when the recording start sensor 10 detects the start of passage of the test target device 200, the channel extraction unit 53 causes the recording data storage unit 51 to record the sound detected by the microphone 20 (step S2).

  Next, the channel extraction unit 53 ends the recording by the recording data storage unit 51 after the time that the first speaker 205a and the second speaker 205b are scheduled to pass under the microphone 20 has elapsed (step S3). Next, the Fourier transform unit 56 performs a short-time Fourier transform on the entire recording data. Thereby, the channel extraction part 53 extracts the time change of the specific frequency (Fp) which is the frequency of a cueing signal (step S4). Next, the channel extraction unit 53 searches for all wide pulse portions equal to the duration of the cue signal from the extracted components of the specific frequency (Fp) (step S5).

  Next, when the narrow-width pulse is detected in the vicinity of Tg1 + Ts_p from the trailing edge of the wide pulse start signal, the channel extracting unit 53 recognizes the portion as the first chirp signal. If the narrow pulse is detected in the vicinity of Tg1 + Ts_e + Tg2 + Ts_p from the trailing edge of the wide pulse cueing signal, the channel extracting unit 53 recognizes the portion as the second chirp signal (step S6).

  Next, the channel extraction unit 53 extracts the amplitude value of the first chirp signal over the entire recording signal, and obtains the passage time TRM at which the envelope of the amplitude value becomes the maximum value ARM. Further, the channel extraction unit 53 extracts the amplitude value of the second chirp signal over the entire recording signal, and obtains the passage time TLM at which the envelope of the amplitude value becomes the maximum value ALM (step S7).

  Next, the determination unit 57 determines whether or not the passage time TRM is earlier than the passage time TLM (step S8). When it is determined as “No” in Step S8, the output unit 58 outputs a signal notifying the abnormality of the wiring (Step S9). When it is determined as “Yes” in Step S9, the waveform extraction unit 54 extracts the first chirp signal near the passage time TRM and extracts the second chirp signal near the passage time TLM (Step S10). For example, the waveform extraction unit 54 extracts a recording signal in a period from the time when the specific frequency (Fp) is detected by the time Ts_p to the start point and the duration until the duration Ts_e elapses. Extract. The waveform extraction unit 54 may extract and average a plurality of first chirp signals near the passage time TRM, and may extract and average a plurality of second chirp signals near the passage time TLM.

  Next, the determination unit 57 determines whether the frequency characteristic of each extracted chirp signal is good or not, and the output unit 58 outputs the determination result (step S11). Next, the determination unit 57 determines whether or not to perform a sound test of the next test target device 200 (step S12). If “Yes” is determined in step S12, the process is executed again from step S2. If it is determined as “No” in step S12, the execution of the flowchart ends.

  According to the present embodiment, a specific frequency component is detected using a sound sensor that moves relative to a test target device equipped with a plurality of sound output devices, and is output in order by each sound output device. Sound source signals can be separated and extracted. Thereby, a sound test can be performed when a test object apparatus carrying a plurality of sound output devices is conveyed. Moreover, since it is not necessary to use a plurality of microphones, it is not necessary to match the characteristics of the plurality of microphones. That is, a sound test with high reproducibility can be performed. Moreover, since it is not necessary to synchronize the sound output by the sound output device and the sound detection by the sound sensor, it is not necessary to provide equipment for synchronization.

  Further, by separating and extracting the sound source signals output in order by each sound output device, it is possible to determine the unbalance of each channel sound source, the wiring change, and the like. In addition, since the time at which each frequency appears in the chirp signal is short, the accuracy in extracting each channel sound source signal is improved by detecting the specific frequency (Fp). Specifically, the time resolution of each channel sound source signal is improved. In this case, the external noise removal process becomes easy. In the above example, a chirp signal is used for evaluation of sound quality characteristics, but the present invention is not limited to this. For example, a signal having a width in frequency can be used for evaluation of sound quality characteristics. If the sound quality characteristic is not evaluated, a single frequency signal may be used for each channel sound source signal.

  In the above example, the number of sound output devices is “2”, but is not limited thereto. In the case of a test target device equipped with three or more sound output devices, sound source signals output by each sound output device can be separated and extracted by causing each sound output device to reproduce sound source signals in order. In the above example, the frequency characteristic is evaluated as the sound quality evaluation. However, in addition to the frequency characteristic, evaluations such as sound cracking and skipping may be performed.

  FIG. 14A is an example of sound source signal data when the sound quality characteristic evaluation is not performed. Referring to FIG. 14A, the sound source signal data may be a single frequency sine wave signal having the same specific frequency (Fp) as the cue signal as the first channel sound source signal and the second channel sound source signal. Good. If the frequencies of the first channel sound source signal and the second channel sound source signal are made the same as the specific frequency (Fp), there is an advantage that component extraction of the specific frequency (Fp) can be performed simultaneously.

  In order to distinguish from the cue signal based on the pulse width of the extracted component of the specific frequency (Fp), the duration Tj of each channel sound source signal is set longer or shorter than the cue signal duration Th. In the example of FIG. 14A, the duration Tj is set longer than the duration Th. When it is necessary to align the duration of each channel sound source signal, it is preferable to provide a cue signal in order to distinguish which channel the sound source signal being reproduced is. This is because by providing the cue signal, it is possible to detect which channel's sound source signal is reproduced following the cue signal. The duration Th and the duration Tj are stored in the sound source information storage unit 52.

  When it is not necessary to align the duration of each channel sound source signal, each speaker can be identified without providing a cue signal with reference to FIG. 14B. This is because it is possible to detect which sound source signal is being reproduced according to the duration of the duration. Even when it is not necessary to align the frequencies of the channel sound source signals, each speaker can be identified without providing a cue signal. This is because it can be detected which sound source signal of which channel is being reproduced according to the detected frequency. In this case, each frequency may be detected as a specific frequency component of the sound source signal of each channel.

  FIG. 15 is a diagram for explaining an example of a flowchart executed when a sound test is performed using the sound source signal of FIG. Hereinafter, the flowchart of FIG. 15 will be described. First, the channel extraction unit 53 reads from the sound source information storage unit 52 the duration Th of the cue signal, the duration Tj of the sound source signal of each channel, and the silence times Tg1, Tg2 (step S21). Next, when the recording start sensor 10 detects the start of passage of the test target device 200, the channel extraction unit 53 causes the recording data storage unit 51 to record the sound detected by the microphone 20 (step S22).

  Next, the channel extraction unit 53 ends the recording by the recording data storage unit 51 after the time that the first speaker 205a and the second speaker 205b are scheduled to pass below the microphone 20 (step S23). Next, the Fourier transform unit 56 performs a short-time Fourier transform on the entire recording data. Thereby, the channel extraction part 53 extracts the time change of the specific frequency (Fp) which is the frequency of a cueing signal (step S24). Next, the channel extraction unit 53 searches for all wide pulse portions equal to the duration Th of the cue signal from the extracted components of the specific frequency (Fp) (step S25).

  Next, when the channel extraction unit 53 detects a pulse having a duration Tj near the end of the silent time Tg1 from the trailing edge of the wide pulse cueing signal, the channel extraction unit 53 recognizes the portion as the first channel sound source signal. Further, when the channel extraction 53 detects a pulse having a duration of Tj near the end of Tg1 + Tj + Tg2 from the trailing edge of the wide pulse cueing signal, the channel extraction 53 recognizes that portion as the second channel sound source signal (step S26). That is, in step S26, the channel extraction unit 53 extracts each channel component.

  Next, the determination unit 57 extracts the amplitude value of the first channel sound source signal over the entire recording signal, and obtains the passage time TRM at which the envelope of the amplitude value becomes the maximum value ARM. Further, the determination unit 57 extracts the amplitude value of the second channel sound source signal over the entire recording signal, and obtains the passage time TLM at which the envelope of the amplitude value becomes the maximum value ALM (step S27).

  Next, the determination unit 57 determines whether or not the passage time TRM is earlier than the passage time TLM (step S28). When it is determined as “No” in Step S8, the output unit 58 outputs a signal notifying the abnormality of the wiring (Step S29). When it is determined as “Yes” in Step S28, the determination unit 57 determines whether or not the maximum value ARM is within a predetermined range (Step S30). When it determines with "No" in step S30, the output part 58 outputs the signal which notifies the volume abnormality of the 1st speaker 205a (step S31). When it determines with "Yes" in step S30, the determination part 57 determines whether the maximum value ALM exists in the predetermined range (step S32). When it determines with "No" in step S32, the output part 58 outputs the signal which notifies the volume abnormality of the 2nd speaker 205b (step S33).

  When it determines with "Yes" in step S32, the output part 58 outputs the signal which notifies the sound volume pass of the 1st speaker 205a and the 2nd speaker 205b (step S34). Next, the determination unit 57 determines whether or not to perform a sound test of the next test target device 200 (step S35). If “Yes” is determined in step S35, the process is executed again from step S22. If it is determined “No” in step S35, the execution of the flowchart ends.

  As described above, if the sound quality characteristic is not evaluated, a single frequency signal may be used as each channel sound source signal. When the sound source signal of FIG. 14B is used, each channel sound source signal may be detected according to the duration of the reproduced sound. Further, when the frequency is different for each channel, each channel sound source signal may be detected according to the frequency of the sound to be reproduced.

  In the first embodiment, a test target device equipped with a plurality of sound output devices has been described. In the second embodiment, a sound test for a test target device 200a including a single sound output device that synthesizes and reproduces a plurality of channel sound sources will be described. FIG. 16A is a block diagram for explaining the configuration of the test target device 200a. Referring to FIG. 16A, a test target device 200a includes a sound source data storage unit 201, a control unit 202, a first D / A converter 203a, a second D / A converter 203b, an adder 207, an output amplifier 204, And a speaker 205.

  The control unit 202 generates a plurality of channels of sound source signals from the sound source signal data stored in the sound source data storage unit 201. In the present embodiment, the control unit 202 generates two types (two channels) of sound source signals, a first channel sound source signal and a second channel sound source signal. The first D / A converter 203a converts the first channel sound source signal into an analog signal, and the second D / A converter 203b converts the second channel sound source signal into an analog signal.

  The adder 207 adds the analog signals output from the first D / A converter 203a and the second D / A converter 203b to each other and outputs the result. The adder 207 outputs an analog signal output from the first D / A converter 203a when only the first D / A converter 203a outputs an analog signal. The adder 207 outputs an analog signal output from the second D / A converter 203b when only the second D / A converter 203b outputs an analog signal. The output amplifier 204 amplifies the analog signal output from the adder 207. The speaker 205 outputs sound according to the amplified analog signal.

  The test apparatus 100 may have the same configuration as that of the first embodiment. FIG. 16B is a schematic diagram for explaining the positional relationship among the recording start sensor 10, the microphone 20, and the test target device 200a.

  Also in this embodiment, when reproducing the sound source signal of FIGS. 6A and 6B, the recording data as shown in FIGS. 7A and 7B is obtained. Therefore, as in the first embodiment, each channel sound source signal can be separated and extracted based on the specific frequency (Fp). In this embodiment, since there is one speaker, unlike the example of FIG. 7A, the difference in the amplitude of each channel sound source signal is reduced.

  FIG. 17 is a diagram for explaining another example of a flowchart executed when a sound test is performed using the sound source signal of FIG. Hereinafter, the flowchart of FIG. 17 will be described. First, the channel extraction unit 53 reads from the sound source information storage unit 52 the duration Th of the cue signal, the duration Tj of the sound source signal of each channel, and the silence times Tg1 and Tg2 (step S41). Next, when the recording start sensor 10 detects the start of passage of the test target device 200, the channel extraction unit 53 causes the recording data storage unit 51 to record the sound detected by the microphone 20 (step S42).

  Next, the channel extraction unit 53 ends the recording by the recording data storage unit 51 after the elapse of time when the speaker 205 is scheduled to pass under the microphone 20 (step S43). Next, the Fourier transform unit 56 performs a short-time Fourier transform on the entire recording data. Thereby, the channel extraction part 53 extracts the time change of the specific frequency (Fp) which is the frequency of a cueing signal (step S44). Next, the channel extraction unit 53 searches for all wide pulse portions equal to the duration Th of the cue signal from the extracted components of the specific frequency (Fp) (step S45).

  Next, when the channel extraction unit 53 detects a pulse having a duration Tj near the end of the silent time Tg1 from the trailing edge of the wide pulse cueing signal, the channel extraction unit 53 recognizes the portion as the first channel sound source signal. Further, when the channel extraction unit 53 detects a pulse having a duration Tj in the vicinity of Tg1 + Tj + Tg2 from the trailing edge of the wide pulse cueing signal, the channel extraction unit 53 recognizes the portion as the second channel sound source signal (step S46).

  Next, the channel extraction unit 53 extracts the amplitude value of the first channel sound source signal over the entire recording signal, and the envelope of the amplitude value obtains the maximum value ARM. Further, the channel extraction unit 53 extracts the amplitude value of the second channel sound source signal over the entire recording signal, and obtains the maximum value ALM from the envelope of the amplitude value (step S47).

  Next, the determination unit 57 determines whether or not the maximum value ARM is within a predetermined range (step S48). By executing step S48, there is no abnormality in the first channel sound source signal generation process, the wiring for transmitting the first channel sound source signal, the path of the first channel signal to the speaker 205 including the first D / A converter 203a, and the like. Can be determined. When it determines with "No" in step S48, the output part 58 outputs the signal which notifies abnormality of a 1st channel (step S49). When it determines with "Yes" in step S48, the determination part 57 determines whether the maximum value ALM exists in the predetermined range (step S50). Whether or not there is any abnormality in the path of the second channel signal to the speaker 205 including the second channel sound source signal generation process, the wiring for transmitting the second channel sound source signal, the second D / A converter 203b, etc. Can be determined. When it determines with "No" in step S50, the output part 58 outputs the signal which notifies abnormality of a 2nd channel (step S51).

  When it determines with "Yes" in step S50, the output part 58 outputs the signal which notifies the pass of a 1st channel and a 2nd channel (step S52). Next, the determination unit 57 determines whether or not to perform a sound test of the next test target device 200 (step S53). If “Yes” is determined in step S53, the process is executed again from step S42. If it is determined “No” in step S53, the execution of the flowchart ends.

  In the present embodiment, each channel that is sequentially output by the sound output device by detecting a specific frequency component using a sound sensor that moves relative to the sound output device that reproduces a plurality of channel sound sources. Signals can be separated and extracted. Thereby, a sound test can be performed when a sound output device for reproducing a multi-channel sound source is transported. Note that the sound quality characteristics of each channel sound source signal can also be evaluated by using the sound source signals shown in FIGS. 6A and 6B.

  In each of the above examples, the control unit 202 generates each channel sound source signal, but is not limited thereto. Each channel sound source signal may be generated by an individual generation circuit or the like. That is, it is only necessary to have a device that generates a plurality of channel sound source signals.

  A recording medium in which a software program for realizing the function of the test apparatus 100 is recorded may be supplied to the test apparatus 100 and the CPU 101 may execute the program. Examples of the storage medium for supplying the program include a CD-ROM, DVD, Blu-ray, or SD card. Also, a dedicated circuit for realizing each unit in FIG. 2B may be used.

  Although the embodiments of the present invention have been described in detail above, the present invention is not limited to such specific embodiments, and various modifications and changes can be made within the scope of the gist of the present invention described in the claims. It can be changed.

DESCRIPTION OF SYMBOLS 10 Recording start sensor 20 Microphone 30 Input amplifier 40 A / D converter 50 Calculation part 51 Recording data storage part 52 Sound source information storage part 53 Channel extraction part 54 Waveform extraction part 55 Noise processing part 56 Fourier transform part 57 Determination part 58 Output part 59 Volume correction unit 100 Test apparatus 101 CPU
102 RAM
DESCRIPTION OF SYMBOLS 103 Memory | storage device 200 Test object apparatus 201 Sound source data storage part 202 Control part 203 D / A converter 204 Output amplifier 205 Speaker 206 Belt conveyor line

Claims (12)

  By detecting a specific frequency component using a sound sensor that moves relative to a test target device equipped with a plurality of sound output devices, sound source signals output in order by the plurality of sound output devices are separated. The test method characterized by extracting as follows.   The test method according to claim 1, wherein one or more of the specific frequency components are included in a sound source signal output by the plurality of sound output devices.   The test method according to claim 2, wherein the sound source signal is a chirp signal.   The test method according to claim 2, wherein the sound source signals output by the plurality of sound output devices have different frequencies.   The test method according to any one of claims 1 to 4, wherein a mounting position of the plurality of sound output devices is determined based on a volume change of the extracted sound source signal.   The test method according to claim 1, wherein the quality of the frequency characteristics obtained by performing a short-time Fourier transform on the extracted sound source signal is determined. A sound sensor that moves relative to a test target device equipped with a plurality of sound output devices;
An extraction unit that separates and extracts sound source signals sequentially output by the plurality of sound output devices by detecting a specific frequency component with respect to the signal detected by the sound sensor. Test equipment.   The test apparatus according to claim 7, wherein at least one of the specific frequency components is included in a sound source signal output from the plurality of sound output apparatuses.   The test apparatus according to claim 8, wherein the sound source signal is a chirp signal.   The test apparatus according to claim 8, wherein the sound source signals output by the plurality of sound output apparatuses have different frequencies.   11. The apparatus according to claim 7, further comprising a first determination unit that determines a mounting position of the plurality of sound output devices based on a volume change of the sound source signal extracted by the extraction unit. The test apparatus described.   12. The apparatus according to claim 7, further comprising a second determination unit that determines whether the frequency characteristics obtained by performing short-time Fourier transform on each sound source signal extracted by the extraction unit is good or bad. The test apparatus according to the item.

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