Patent Translate Powered by EPO and Google Notice This translation is machine-generated. It cannot be guaranteed that it is intelligible, accurate, complete, reliable or fit for specific purposes. Critical decisions, such as commercially relevant or financial decisions, should not be based on machine-translation output. DESCRIPTION JP2006313953 PROBLEM TO BE SOLVED: To provide an automatic sound field correction system capable of accurately measuring and correcting delay without requiring complicated processing for all speakers having different reproduction bands. SOLUTION: An automatic sound field correction system according to the present invention comprises a sound field measurement device 1, an amplifier 2, a speaker 3, a microphone 4 and a microphone amplifier 5. The sound field measurement apparatus 1 generates different pulse signals according to the band of the speaker to be measured, and after amplification by the amplifier 2, the speaker 3 reproduces. On the other hand, when the microphone 4 installed at the listening position records a pulse signal, it is amplified by the microphone amplifier 5 and then input to the sound field measurement device 1. The sound field measurement device 1 calculates the delay value to each speaker from the input data, and applies a correction. By using the present invention, it is possible to correct the delay accurately without the need for complicated processing such as DFT. [Selected figure] Figure 1 Automatic sound field correction system, automatic sound field correction method and sound field measurement device [0001] The present invention relates to an automatic sound field correction system, an automatic sound field correction method, and a sound field measurement device that automatically corrects a deviation of a sound field generated due to a difference in distance from each speaker to a listening position in an audio system. [0002] 09-05-2019 1 Conventionally, in an audio system, it is difficult to output the sound of all the required frequency bands with a single speaker, so the sound to be reproduced by the band limiting filter is divided by the frequency band, and the divided sounds are used for the bass The system uses a separate speaker such as for the middle range and high range. Also, in order to give a three-dimensional effect to the sound to be reproduced, speakers of two channels on the left and right are used, or a surround speaker is provided at the back of the viewer. Here, in order for the listener to listen to the sounds independently generated from the respective speakers without discomfort, it is necessary that all the sounds from the respective speakers reach the listening position at the same time. [0003] Patent documents 1 to 6 are known as reference documents of prior art related to the present application. Patent No. 2725838 Patent No. 3148060 Japanese Patent Laid-Open No. 07212896 Japanese Patent Laid-Open No. 11-262081 Japanese Patent Laid-Open No. 11-258034 Japanese Patent Laid-Open No. 2001-224100 [0004] However, since the distance from each speaker to the listening position is not necessarily constant, it is necessary to measure the distance from each speaker to the listening position and to delay the output sound according to the measured distance. Here, as a conventional method of measuring the distance to the listening position, a method using TSP (Time Stretched Pulse) or white noise is introduced, but complex processing such as DFT (Discrete Fourier Transform) is required. Also, in the process of the process, a lot of resources such as memory were required. [0005] Another known method is to measure the distance to the listening position using a pulse signal specialized for the low-range woofer, for example, in a multi-way speaker combining a low-range woofer and a high-range tweeter. It is done. However, in the case of in-vehicle multi-way speakers in particular, the positions of the woofer and tweeter are often separated, so in delay 09-05-2019 2 correction using a pulse signal specialized for woofer, listening from the distance from the woofer to the listening position and tweeter It is not possible to correct the difference with the distance to the position. [0006] However, if the same pulse signal is output from the woofer and the tweeter in order to simultaneously measure the listening position of the woofer and the tweeter, the S / N ratio (Signal to Noise Ratio) is obtained because of the difference in the reproduction band of the speaker Is degraded. This is because, for example, a pulse signal specialized for the woofer contains many low frequency components so that a large S / N ratio can be obtained in the distance measurement of the woofer, but when this pulse signal is passed through a tweeter for high frequency range, the low frequency component Is largely attenuated by the band limiting filter. As a result, the output power of the pulse signal decreases, the S / N ratio decreases, and not only does the distance measurement accuracy deteriorate, but also in an environment where a noise source such as a car interior is near, the pulse signal Being buried in the noise source, it becomes impossible to measure the distance. [0007] The present invention has been made in consideration of the above circumstances, and an object thereof is to provide an automatic sound capable of accurately measuring and correcting delay for all speakers having different reproduction bands without requiring complicated processing. It is in providing a field correction system. [0008] The present invention has been made to solve the above-mentioned problems, and the invention according to claim 1 comprises a plurality of speakers having different characteristics, and a sound field forming a sound field at a listening position by the plurality of speakers. An automatic sound field correction system in a forming apparatus, comprising: pulse signal generation means for generating different pulse signals according to the characteristics of the plurality of speakers; and one of the plurality of speakers selected from the plurality of speakers; Speaker selection means for applying a pulse signal generated according to the characteristics of the one speaker by the generation means to the one speaker, a microphone installed at the listening position, an output timing of the pulse signal, and an output signal of the microphone Delay time measuring means for measuring the sound wave delay time between the one speaker and the listening position based on An automatic sound field correcting system characterized by comprising a delay time adjustment means for adjusting the delay time of the signal applied to 09-05-2019 3 said first speaker in accordance with a constant result. [0009] The invention according to claim 2 is the invention according to claim 1, wherein the pulse signal generation means generates pulse signals for respective bands for each of a plurality of bands. And pulse signal selection means for selecting any one of the pulse signals for each band output from the pulse signal generation means for each band according to the characteristics of the one speaker. [0010] Also, in the invention according to claim 3, in the invention according to claim 1 or 2, the delay time calculation means determines that the output signal of the microphone is the first threshold from the output start time of the pulse signal. It is characterized in that a delay time is obtained by calculating a time until it exceeds and subtracting a correction time preset for the pulse signal from the calculated time. [0011] The invention according to a fourth aspect is the invention according to any one of the first to third aspects, wherein the microphone outputs no sound during a predetermined time before the pulse signal is output to the speaker. It is characterized by comprising: DC offset calculation means for calculating a DC offset from the signal; and DC offset subtraction means for subtracting the calculated DC offset from the output signal of the microphone. [0012] In the invention according to claim 5, in the invention according to any one of claims 1 to 4, a peak of an output signal of the microphone after the pulse signal is applied to the speaker of the one. Peak detection means for detecting the level; noise detection means for detecting the level of the output signal of the microphone before the pulse signal is applied to the one speaker; detection results of the peak detection means and the noise detection means Signal-to-noise ratio calculation means for calculating the ratio of detection results and whether or not the speaker of 1 is connected based on whether or not the calculation result of the signal-to-noise ratio calculation means is greater than a second threshold value And a wire connection determination means for determining. [0013] Also, in the invention according to claim 6, in the invention according to claim 5, it is necessary 09-05-2019 4 to measure the sound wave delay time based on whether the calculation result of the signal to noise ratio calculation means is larger than a third threshold. A signal-to-noise ratio determination unit is included to determine whether the signal-to-noise ratio is maintained. [0014] The invention according to claim 7 comprises pulse signal generating means for generating pulse signals having different characteristics, and pulse signal output means for selecting the characteristics of 1 and outputting the pulse signal generated by the pulse signal generating means; A response input means for inputting a response to the output pulse signal, a delay time from an output of the pulse signal to an input at the response input means based on an output timing of the pulse signal and an input signal of the response input means And a delay time measuring means for measuring [0015] The invention according to claim 8 is an automatic sound field correction method in a sound field forming apparatus including a plurality of speakers having different characteristics, and forming a sound field at a listening position by the plurality of speakers, the plurality of speakers A first process of selecting one of the speakers, a second process of generating a pulse signal according to the characteristics of the selected one speaker, and adding the pulse signal to the one speaker, and an output timing of the pulse signal A third process of measuring an acoustic wave delay time between the one speaker and the listening position based on an output signal of the microphone installed at the listening position, and the first process according to the measurement result of the acoustic wave delay time An automatic sound field comprising: a fourth process of adjusting a delay time of a signal applied to a speaker; and performing the first to fourth processes on all the plurality of speakers A positive way. [0016] According to the present invention, a noise source such as a vehicle interior is located close to each other to measure the distance to the listening position by outputting pulse signals from the speakers having different reproduction bands in accordance with the reproduction band of the speakers. It is possible to make a good measurement of S / N ratio also in various environments. In addition, it can be easily realized with a small number of resources without requiring complicated arithmetic processing such as DFT. 09-05-2019 5 [0017] Hereinafter, embodiments of the present invention will be described with reference to the drawings. FIG. 1 is a block diagram showing the configuration of an automatic sound field correction system according to an embodiment of the present invention, and FIG. 2 is a block showing a method for outputting a pulse signal from a sound field measurement apparatus according to an embodiment of the present invention FIG. FIG. 3 is a layout diagram showing the layout of speakers in the room. [0018] In FIG. 1, a sound field measurement device 1 is a device for measuring a sound field, and is made into one chip. The microcomputer 11 of the sound field measurement device 1 is responsible for main control in the sound field measurement device 1. A DSP (Digital Signal Processing) unit 12 receives a control signal from the microcomputer 11 and performs digital signal processing. A D / A converter (Digital-to-Analog Converter) 13 receives a digital signal from the DSP unit 12, converts it into an analog signal, and outputs the analog signal to the amplifier 2. An A / D converter (Analog-to-Digital Converter) 14 receives an analog signal from the microphone amplifier 5, converts it into a digital signal, and outputs the digital signal to the DSP unit 12. 09-05-2019 6 [0019] The amplifier 2 amplifies and outputs an input audio signal. The speaker 3 reproduces the audio signal input from the amplifier 2 as sound. A microphone (microphone) 4 is attached to the listening position and collects the sound reproduced from the speaker 3. The microphone amplifier 5 amplifies an audio signal input from the microphone 4 and outputs the amplified signal to the A / D converter 14. [0020] The audio input unit 6 is for inputting an audio signal from an external audio reproduction device, and the audio decoding unit 7 is for decoding the audio signal inputted by the audio input unit 6. The post processor unit 8 adds various acoustic effects to the audio signal decoded by the audio decoding unit 7. The delay unit (delay time adjustment means) 9 delays the audio signal to be transmitted to each speaker according to the instruction from the microcomputer 11. In the embodiment of the present invention, the one-chip sound field measurement apparatus 1, the amplifier 2, the microphone amplifier 5, the audio input unit 6, the audio decoding unit 7, the post processor unit 8, and the delay unit 9 It is realized by being incorporated in the device 10). Although only one amplifier 2, one speaker 3 and one delay unit 9 are shown, in actuality there are a plurality of channels. [0021] FIG. 2 is a view showing a method in which the sound field measurement apparatus 1 of FIG. 1 outputs a pulse signal for sound field measurement, and shows functions performed by the microcomputer 11 and the DSP unit 12. In FIG. 2, the low frequency pulse signal generation unit (pulse signal generation unit for each band) 101 of the sound field measurement device 1 generates a pulse signal for output to a speaker that handles a low frequency region such as a woofer or a subwoofer. The pulse signal generated here is a Raised Cosine type pulse signal and contains many low frequency components. The high-frequency pulse signal generation unit (pulse signal generation means for each band) 102 generates a pulse signal for output to a speaker that handles a high-frequency range such as tweeter, and the pulse signal generated 09-05-2019 7 here is Raised Cosine Is a type of pulse signal and contains many high frequency components. [0022] The pulse selection unit (pulse signal selection unit) 103 selects which of the low-frequency pulse signal and the high-frequency pulse signal to be output, and the switching signal generation unit 104 selects the type of pulse signal to be output. Is generated to designate the pulse select section. The level adjustment unit 105 adjusts the power of the pulse signal to be output. [0023] The speaker selection unit (speaker selection unit) 106 selects which speaker the pulse signal is to be output to, and the output speaker selection signal generation unit 107 uses the speaker selection unit 106 to select the type of speaker outputting the pulse signal. It generates an output speaker selection signal for designating. The CH level adjustment unit 108 adjusts the power of the pulse signal to be output for each speaker. The amplifier 2 amplifies the power of the pulse signal output from the CH level adjustment unit 108, and the speaker unit 3 is a plurality of speakers constituting an audio system. [0024] FIG. 3 is a diagram showing an example of arranging eight types of speakers shown in the speaker unit 3 in the listening room 100. As shown in FIG. In FIG. 3, the installation place of the microphone 4 is a position where the listener listens to the audio. The R tweeter 301 outputs the high range in the speaker disposed on the front right side of the listener, and the R woofer 302 outputs the low range of the same speaker. The L tweeter 303 outputs a high range in a speaker disposed on the front left side of the listener, and the L woofer 304 outputs a low range of the same speaker. The center 305 is a speaker disposed in the front center of the listener. The R surround 306 is a speaker disposed on the rear right side of the listener, and the L surround 307 is a speaker disposed on the rear left side of the listener. The subwoofer 308 is a speaker that is disposed diagonally forward to the right of the listener and outputs a bass range further than the woofer. 09-05-2019 8 [0025] Next, the operation of the above-described embodiment will be described with reference to FIGS. 4 to 8. FIG. 4 is a flow chart showing the procedure performed by the microcomputer 11 and the DSP unit 12 when performing sound field measurement, and FIG. 5 is a flow chart showing the procedure in the subroutine shown in step S409 of FIG. Is a flowchart showing the procedure in the subroutine shown in step S410 of FIG. In FIG. 4, first, the microcomputer 11 selects a speaker to be measured, and the output speaker selection signal generation unit 107 outputs an output speaker selection signal to the speaker selection unit 106 (step S401). The speaker selection unit 106 sets a speaker to be output based on the input output speaker selection signal. [0026] When the output speaker is determined, the microcomputer 11 selects the type of pulse signal from the band handled by the speaker, and the switching signal generator 104 outputs a switching signal to the pulse selector 103 (step S402). The pulse selector 103 selects an output pulse based on the input switching signal 104. Here, a procedure for outputting a low frequency pulse signal will be described. [0027] When the output destination speaker and the output pulse are determined, the DSP unit 12 starts measuring the distance to the speaker. In the following, the sampling interval is t seconds, and time is represented by the number of times of sampling. For example, the delay time of 10 samples represents the time required to perform sampling 10 times, ie, 10 t seconds. In the distance measurement, as shown in the graph on the upper side of FIG. 7, the signal input from the microphone 4 through the microphone amplifier 5 is stored in the memory for M samples in a state where no pulse signal is output (step S403 in FIG. 4). ). After the M samples have elapsed, a pulse signal is output (step S404), and the N-M samples are stored in the memory (step S405). At this time, the signal input by the microphone 4 is a graph as shown on the lower side of FIG. [0028] The series of measurements (steps S403 to S405) are repeated the number of times set in 09-05-2019 9 advance by the microcomputer 11 to perform synchronous addition (step S406). The synchronous addition is to add the data obtained by repeatedly performing the same measurement while aligning the time axis, and taking the average to suppress the influence of noise and improve the S / N ratio. When the synchronous addition is completed, the average value of the measurement results of the first half M samples is determined (DC offset calculating means), and the average value of the first half M samples determined from the data of the average values of all N samples is subtracted (DC offset subtracting means, Step S407). This is to remove the DC offset (DC offset) generated at the time of A / D conversion from the measurement result. [0029] The maximum value is detected from the data of the first half M samples after removing the DC offset, and the DSP unit 12 stores the larger one of the maximum value and the preset value of NoiseMin as the noise level (noise detection means, Step S408). Subsequently, a peak level is detected from the latter half NM samples (peak detection means), and wire connection determination and S / N ratio determination are performed (step S409). Next, the procedure of step S409 (subroutine) will be specifically described with reference to FIG. [0030] The maximum value is detected from the second half NM samples after removing the DC offset (step S501), and the DSP unit 12 stores the detected maximum value as a peak level in the memory and the register (step S502). Next, determination of unconnected and determination of S / N ratio are performed from the stored noise level and peak level. In the determination of unconnected, the ratio of the peak level to the noise level is calculated (signal-to-noise ratio calculating means), and it is confirmed whether it falls below a preset threshold 1 (connection determining means, step S503). For example, when the threshold 1 is 1.4, if the peak level is smaller than 1.4 times the noise level, it is determined as unconnected. [0031] If it is determined that the connection is not yet made (step S503: No), the S / N ratio is subsequently determined. Here, the threshold for delay measurement (threshold 2) and the threshold for polarity determination (threshold 3) are set in advance, and the ratio of peak level 09-05-2019 10 to noise level is smaller than the ratio of threshold 2 to threshold 3 or not Make sure. Here, threshold 2 and threshold 3 are variables. Assuming that threshold 2 is 2.0 and threshold 3 is 0.4, if the peak level is smaller than 5 times the noise level, it is determined that the S / N ratio is insufficient. [0032] If it is determined in step S503 that the value is lower than the threshold 1, the DSP unit 12 outputs an error indicating that the connection is not yet made. If it is determined in step S504 that the S / N ratio is insufficient, an error to that effect is output. The content of the error output from the DSP unit 12 is displayed on the screen (not shown) of the amplifier device 10 through the microcomputer 11. [0033] Returning to FIG. 4, when the wire connection determination and the S / N ratio determination are completed, a delay value is finally obtained (delay time measuring means, step S410). The procedure of step S410 (subroutine) will be specifically described with reference to FIG. In the present embodiment, NoiseMin is set to a level of −42 dB, and a value obtained by multiplying the noise level stored in step S409 by the threshold 2 is set as the threshold 4. In the upper diagram in FIG. 7, the time from when the pulse signal is output after M sampling until the level (amplitude) of the output pulse signal exceeds threshold 4 for the first time is stored as Δ sample, and correction is performed in the later procedure Used when. Here, the Δ sample varies depending on the band of the pulse signal to be used. In this embodiment, in the pulse for each band, the time (the number of samples) from the output start point of the pulse to the peak is made to coincide, so for example, for the high frequency band as shown in FIG. The Δ sample has a large value with the pulse signal of [0034] In the procedure for obtaining the delay value, in the lower graph of FIG. 7 representing the level of the signal input by the microphone 4, first, the DSP unit 12 detects one sample at a time from the start point (after M samples) where the pulse signal is output It is determined whether the level exceeds the threshold 4 (step S601 in FIG. 6). If the threshold value 4 is exceeded (step S602: YES), the Δ sample is subtracted from the elapsed time (D sample) from the output of the 09-05-2019 11 pulse signal to obtain a delay value (step S603). Thereafter, returning to FIG. 4, the procedure of sound field measurement for one speaker is completed. On the other hand, when the threshold value 4 is not exceeded (step S602: No), it is determined whether the determination of the prescribed number of samples is completed (step S604). Here, if the determination for the prescribed samples is completed (step S604: Yes), the DSP unit 12 outputs an error indicating that the delay value could not be measured, assuming that no sample exceeding the threshold 4 is found (step S605), Display on the screen (not shown) of the amplifier device 10 via the microcomputer 11. If the determination for the prescribed sample is not completed (step S604: YES), the process returns to step S601 to determine whether the threshold value 4 is exceeded for the next sample. [0035] Essentially, the delay value to be determined is the time from the start point (after M samples) at which the pulse signal was output to the start point at which the pulse signal was detected by the microphone 4. However, as shown in the lower diagram of FIG. It is difficult to determine the start point at which the pulse signal is detected due to the influence of Therefore, a method is employed in which D samples are measured based on the threshold value 4 and then correction is performed (Δ sample is subtracted) according to the waveform of the pulse signal. The number of specified samples in step S604 is an appropriate number equal to or less than the number of M-N samples. [0036] Thus, the measurement for one speaker is completed, and thereafter, the procedure of steps S401 to S410 is repeatedly performed for each speaker to obtain delay values for all the speakers. When the delay values for all the speakers are obtained, the microcomputer 11 calculates the amount of delay given to each speaker so as to fit to the speaker having the maximum delay value. For example, the delay value for R tweeter 301 is 108 samples, the delay value for R woofer 302 is 264 samples, the delay value for L tweeter 303 is 152 samples, and the delay value for L woofer 304 is 193 samples. Consider the case of correcting the delay amount. At this time, the maximum value of the delay value is 264 samples of R woofer 302, the delay amount of 156 samples for R tweeter 301, the delay amount of 0 samples for R woofer 302, and the delay amount of 112 samples for L tweeter 303 The delay amount of 71 samples is set to L woofer 304. 09-05-2019 12 [0037] The calculated delay amount is output from the microcomputer 11 to the delay device 9 of each speaker, and the delay device 9 receiving the delay amount gives the audio signal a delay amount and outputs it to the amplifier 2. In the above example, the audio signal reproduced by R tweeter 301 has a delay of 156 samples, the audio signal reproduced by R woofer 302 has a delay of 0 samples, and the audio signal reproduced by L tweeter 303 has 112 samples The delay 9 gives a delay of 71 samples to the audio signal reproduced by the L woofer 304. Thus, the arrival time from each speaker to the microphone 4 is matched by adding a delay when outputting an audio signal. [0038] The sound field automatic correction system according to the embodiment of the present invention adopts a method using a single pulse signal, and does not require complicated processing such as DFT. This not only makes it possible to shorten the processing time required to measure the delay value, but also has the advantage that it can be realized even if the capacity of the memory required by the sound field measurement device 1 is small. [0039] As mentioned above, although the embodiment of the present invention has been described in detail, the specific configuration is not limited to the present embodiment, and design changes and the like within the scope of the present invention are also included. For example, although the threshold value 4 is calculated when the maximum value of noise obtained in step S408 exceeds NoiseMin in the present invention, the threshold value 4 may always be varied according to the maximum value of noise. [0040] The present invention is suitable for use in an automatic sound field correction system which automatically corrects the bias of the sound field caused by the difference in the distance from each speaker to the listening position in an audio system. [0041] 09-05-2019 13 FIG. 1 is a configuration diagram showing a configuration of an automatic sound field correction system according to an embodiment of the present invention. It is the block diagram which showed the method of outputting a pulse signal from the sound field measurement apparatus of FIG. It is a layout showing the arrangement of the speaker in the room. It is the flowchart which showed the procedure performed when the microcomputer 11 of FIG. 1 performs a sound field measurement. It is the flowchart which showed the procedure in the subroutine shown by FIG.4 S409. It is the flowchart which showed the procedure in the subroutine shown by FIG.4 S410. It is the graph which showed the pulse signal for low-pass outputted from speaker 3 of Drawing 1, and the pulse signal which microphone 4 inputs. It is the graph showing the pulse signal for high frequencies output from the speaker 3 of FIG. Explanation of sign [0042] DESCRIPTION OF SYMBOLS 1 ... sound field measurement apparatus, 2 ... amplifier, 3 ... speaker, 4 ... microphone (microphone), 5 ... microphone amplifier, 9 ... delay device (delay time adjustment means), 10 ... amplifier apparatus, 11 ... microcomputer, 12 ... DSP unit, 13 ... D / A converter, 14 ... A / D converter, 101 ... pulse signal generator for low frequency band (pulse signal generator for each band), 102 ... pulse signal generator for high frequency band (pulse for each band Signal generation means), 103 ... pulse select part (pulse signal selection means), 104 ... switching signal generation part, 105 ... level adjustment part, 106 ... speaker selection part (speaker selection means), 107 ... output speaker selection signal generation part, 108 ... CH level adjustment section 09-05-2019 14
© Copyright 2021 DropDoc