Patent Application
20080144839
The present invention relates to a characteristics measurement, i.e., measuring of characteristics subject to a measurement in a certain environment.
In a specific environment, various kinds of characteristics subjected to measurements are measured. As an example of the characteristics measurement, there are a system for measuring sound characteristics in a certain sound space and a system for measuring transmission characteristics of a light and an electric wave in a certain environment.
For example, in an audio system including plural speakers and providing a high-grade sound space, it is necessary to automatically create an appropriate sound space with the presence. Namely, even though a listener operates the audio system in order to obtain the appropriate sound space, it is extremely difficult for him or her to appropriately control phase characteristics, frequency characteristics and a sound pressure level of sounds reproduced by the plural speakers. Therefore, it becomes necessary to automatically correct sound field characteristics in the audio system.
Conventionally, as an automatic sound field correcting system of this kind, there is known a system disclosed in Patent Reference-1. In this system, a test signal outputted from a speaker is collected, and frequency characteristics thereof are analyzed, for each signal transmission path corresponding to plural channels. Then, a coefficient of an equalizer arranged in the signal transmission path is set. Thereby, the frequency characteristics in each signal transmission path are desirably corrected.
Additionally, a signal delay time of each signal transmission path corresponding to the plural channels is measured, and the signal delay characteristics of each transmission path are adjusted. In the normal signal delay time measurement, a processor in an automatic sound field correcting system outputs a measurement pulse, and at the same time, the processor starts capturing microphone input. Then, the time until the level of the microphone input becomes larger than a predetermined threshold for the first time is determined as the signal delay time.
As for the above-mentioned characteristics measurement, there is known such a technique that the same measurement is executed for plural times and measurement results are obtained. Namely, the measurement is executed for the plural times, because of a cause existing in an environment in which the measurement is executed and causing a variation of the measurement result, i.e., for the purpose of removing an influence of a noise in the measurement environment and improving the measurement accuracy. In this case, it is general that the number of times of measurement is a fixed number predetermined based on the noise state in the environment.
Patent Reference-1: Japanese Patent Application Laid-open under No. 2002-330499
However, in the case of fixing the number of times of measurement, the number of times of measurement has to be determined by assuming a case of the worst noise state (e.g., a case of a bad S/N state) in the environment and considering completion of the measurements in an actual time period. Hence, even when the actual environment is better than the worst noise state, the measurement is executed for the number of times of measurement, which is determined in correspondence with the worst noise state. As a result, it problematically takes longer time than needed to execute the measurement. Meanwhile, in such a case that the noise state better than the worst noise state is assumed and the smaller number of times of measurement is set in order to shorten the measurement time, if the noise state in the actual environment is worse than assumed, it problematically becomes impossible to obtain the accurate measurement result.
The present invention has been achieved in order to solve the above problems. It is an object of this invention to provide a characteristics measurement device and a program capable of obtaining a measurement result with high accuracy in the minimum number of times of measurement, in accordance with a noise state in an environment in which the measurement is executed.
According to one aspect of the present invention, there is provided a characteristics measurement device which measures characteristics subjected to a measurement, including: a noise level measurement unit which measures a noise level in an environment subjected to the measurement; a noise state determination unit which determines a noise state in the environment, based on the noise level; a measurement number determination unit which determines a number of times of measurement, based on the noise state; and a characteristics measurement unit which measures the characteristics subjected to the measurement for the number of times of measurement, and executes synchronized addition of measurement results to output the measurement results.
The above characteristics measurement device is applicable to various kinds of measurement devices for measuring the characteristics subjected to the measurement in the certain environment. In addition, the above characteristics measurement device measures the noise level in the environment, and determines the noise state based on the obtained noise level. Then, the characteristics measurement device determines the number of times of measurement of the characteristics based on the noise state, and executes synchronized addition of the characteristics obtained by the plural measurements to output the characteristics. Thus, in such a case that the noise state in the environment in which the measurement is executed is preferable, the measurement is completed in the minimum number of times of measurement. Additionally, in such a case that the noise state in the environment is not preferable, the plural measurements are executed in order to obtain a desired noise state (e.g., S/N), and the results are synchronized and added. Since the synchronized addition is repeated and the influence of the noise is reduced, the measurement results with high accuracy can be obtained.
In a manner, the above characteristics measurement device may further include a signal level measurement unit which measures the signal level subjected to the measurement in the environment, and the noise state determination unit may determine the noise state, based on the signal level and the noise level. In this manner, since the noise state (e.g., S/N) is determined with using the signal level subjected to the measurement in the environment in which the measurement is executed, it becomes possible to determine the accurate noise state in the environment.
In a preferred example, the noise level measurement unit may measure the noise level prior to the measurement of the characteristics subjected to the measurement. The noise level measurement unit may measure the noise level during the measurement of the characteristics subjected to the measurement. Moreover, the noise level measurement unit may measure the noise level prior to the measurement of the characteristics subjected to the measurement, and may measure the noise level during the measurement of the characteristics subjected to the measurement. The noise state determination unit may determine the noise state, based on a largest noise level which is measured.
In another manner of the above characteristics measurement device, the measurement number determination unit may increase the number of times of measurement, as the noise state becomes insufficient. Thus, by the effect of the synchronized addition, the influence of the noise in the measurement result can be reduced, and the measurement result with the high accuracy can be obtained.
In still another manner, the above characteristics measurement device may further include a correlation determination unit which determines a correlation of the plural measurement results, and the measurement number determination unit may increase the number of times of measurement, when the correlation is smaller than a predetermined reference. In the environment in which the measurement is executed, an unexpected noise other than an ordinary noise may occur. If the unexpected noise occurs, the measurement accuracy extremely becomes low. Therefore, when the correlation of the plural measurement results is low, it is assumed that the unexpected noise occurs. By increasing the number of times of measurement, the influence of the unexpected noise can be removed.
According to another aspect of the present invention, there is provided a characteristics measurement device which measures characteristics subjected to a measurement, including: a characteristics measurement unit which measures the characteristics subjected to the measurement for a number of plural measurements and executes synchronized addition of measurement results to output the measurement results; a correlation determination unit which determines a correlation of the plural measurement results; and a measurement number determination unit which determines the number of times of measurement, based on a determination result of the correlation.
The above characteristics measurement device is applicable to various kinds of measurement devices which measures the characteristics subjected to the measurement in the environment. The above characteristics measurement device measures the characteristics subjected to the measurement in the number of plural measurements, and executes the synchronized addition of the measurement results to output the measurement results. Then, the characteristics measurement device measures the noise level in the environment, and determines the noise state based on the obtained noise level. In the environment in which the measurement is executed, the unexpected noise other than the ordinary noise may occur. If the unexpected noise occurs, the measurement accuracy extremely becomes low. Therefore, when the correlation of the plural measurement results is low, it is assumed that the unexpected noise occurs. By increasing the number of times of measurement, the influence of the unexpected noise can be removed.
In a preferred example of the above characteristics measurement device, the characteristics subjected to the measurement may be any one of a sound characteristic, a light transmission characteristic, a wave transmission characteristic and an electric circuit characteristic. In addition, the sound characteristics may be any one of a signal delay characteristic, a sound pressure level characteristic, a frequency characteristic and a speaker characteristic in a sound space.
According to still another aspect of the present invention, there is provided a characteristics measurement program executed on a computer and measuring characteristics subjected to a measurement, making the computer function as: a noise level measurement unit which measures a noise level in an environment subjected to a measurement; a noise state determination unit which determines a noise state in the environment, based on the noise level; a measurement number determination unit which determines a number of times of measurement, based on the noise state; and a characteristics measurement unit which measures the characteristics subjected to the measurement for the number of times of measurement, and executes synchronized addition of measurement results to output the measurement results. By executing the program on the computer, the above characteristics measurement device can be realized.
According to still another aspect of the present invention, there is provided a characteristics measurement program executed on a computer and measuring characteristics subjected to a measurement, making the computer function as: a characteristics measurement unit which measures characteristics subjected to the measurement for a number of plural measurements, and executes synchronized addition of measurement results to output the measurement results; a correlation determination unit which determines a correlation of the plural measurement results; and a measurement number determination unit which determines the number of times of measurement, based on the determination result of the correlation. By executing the program on the computer, the above characteristics measurement device can be realized.
The preferred embodiment of the present invention will now be described below with reference to the attached drawings. Hereinafter, a description will be given of such a case that a characteristics measurement technique according to the present invention is applied to the signal delay time measurement in the sound space.
First, the description will be given of a basic principle of a signal delay time measurement according to the present invention.
The measurement signal generator 3 generates the pulse signal (hereafter, referred to as “measurement pulse signal”) as a measurement signal 211, and supplies it to the signal processing circuit 2. The measurement pulse signal can be stored in a memory in the measurement signal generator 3 as a digital signal. The signal processing circuit 2 transmits the measurement pulse signal 211 to the D/A converter 4. The D/A converter 4 converts the measurement pulse signal 211 to an analog measurement pulse signal 212, and supplies it to the speaker 6. The speaker 6 outputs a measurement pulse sound 35 corresponding to the measurement pulse signal 212 to the sound space 260 as the measurement signal sound.
The microphone 8 collects the measurement pulse sound 35 in the sound space 260, and transmits it to the A/D converter 10 as an analog response signal 213. The response signal 213 includes a response component of the sound space 260 to the measurement pulse signal 35. The A/D converter 10 converts the response signal 213 to a digital response signal 214, and supplies it to the signal processing circuit 2. The signal processing circuit 2 calculates a signal delay time Td in the sound space 260 by comparing the response signal 214 with a predetermined threshold.
As understood from
Therefore, even if the sound delay time Tsp is zero (i.e., in a state that the speaker 6 and the microphone 8 are close to each other), since the in-device delay time Tp exists, the signal delay time Td does not become zero. In other words, in a period corresponding to the in-device delay time Tp from the timing at which the signal processing circuit 2 starts outputting the measurement pulse signal 211, the response signal 214 cannot theoretically reach the signal processing circuit 2. Namely, the response signal cannot reach the signal processing circuit 2 in a period (hereafter, referred to as “no-response period”) corresponding to the in-device delay time Tp after the outputting of the measurement pulse signal 211.
As shown in
“Synchronized addition” means that the plural signals are added with maintaining phase information. If the synchronized addition is executed for the plural times, since the phases are same, the number of signal components included in the response signal 214 increases, e.g., twice in the two measurements, three times in the three measurements, and n times in the n measurements. Meanwhile, though the absolute amount of noise components included in the response signal 214 also increases by the plural measurements, the absolute amount increases by √{square root over (2)} times in the two measurements, by √{square root over (3)} times in the three measurements, and by √{square root over (n)} times in the n measurements. Hence, as the number of synchronized additions increases, the ratio between the increase of the noise component and the increase of the signal component becomes small, and thus the S/N is improved.
The actual synchronized addition process is executed by a method explained below, for example. When the number of synchronized additions is n times, the synchronized addition data buffer 231 stores the 1/n data of the response signal 214 obtained from the microphone input buffer 232 for each time. Hence, when the n measurements are completed, the response signal data after the n synchronized additions is stored in the synchronized addition data buffer 231. The synchronized addition data buffer 231 may add the data itself of the response signal 214 for each time, instead of adding the 1/n response signal data for each time, and may execute the process of calculate 1/n of the added result at the time of completing of the n-th measurement. Then, the synchronized addition data buffer 231 supplies the response signal data after the synchronized addition to the switch 235.
Returning to
The correlation determination unit 234 receives the response signal stored in the microphone input buffer 232 as a signal 218, and receives the response signal stored in the synchronized addition data buffer 231 as a signal 219. Then, the correlation determination unit 234 determines the correlation between the signal 218 and the signal 219. When the correlation is smaller than a predetermined reference, the correlation determination unit 234 increases the number of times of measurement. The correlation determination unit 234 has a function to detect the unexpected noise included in the response signal 214. FIG. 4C shows an example of a wave form of the response signal 214 including the unexpected noise 96. When the level of the normal response signal 214 becomes larger than the predetermined threshold as shown in
As one of the concrete determination methods of the correlation, there is a method of calculating correlation values of the response signals 214 shown in
Next, a description will be given of the signal delay measurement unit 2b. The response signal data 215 after the synchronized addition, which is supplied from the synchronized addition data buffer 231 via the switch 235, is inputted into the differentiating circuit 251. The differentiating circuit 251 differentiates the response signal data 215, and calculates the absolute value (ABS) to supply it to the comparator 252.
A background noise measurement unit 253 detects a background noise level from the response signal 214 in a background noise measurement period Tm, which will be described later, and supplies a largest level value thereof to a threshold determination unit 254. The threshold determination unit 254 determines a threshold TH larger than the largest level value of the background noise by a predetermined value, and inputs it to the comparator 252.
A memory 255 stores the in-device delay time Tp, and inputs it to the comparator 252. The comparator 252 compares a differentiating signal of the response signal inputted from the differentiating circuit 251 with the threshold inputted from the threshold determination unit 254, and calculates the signal delay time Td. However, the comparator 252 does not perform the comparison processing of a differentiating value of the response signal and the threshold TH in the no-response period corresponding to the above-mentioned in-device delay time Tp from the timing at which the signal processing circuit 2 starts outputting the measurement signal 211, on the basis of the in-device delay time Tp supplied from the memory 255.
Next, the description will be given of a measurement in the background noise measurement unit 253. As described above, the response of the measurement pulse sound cannot arrive during the period corresponding to the in-device delay time Tp from the time 0 at which the signal processing circuit 2 starts outputting the measurement pulse sound, and the response signal can arrive immediately after the period. Thus, since the background noise level immediately before the execution of the comparison processing of the response signal can be obtained in the period, the period can be quite preferred as a period for detecting the background noise level, which is used to determine the threshold TH. The background noise measurement unit 253 measures the background noise level in the period corresponding to the in-device delay time Tp from the time 0, and based on the level, the threshold determination unit 254 determines the threshold TH used by the comparator 252 in the comparison processing immediately after the measurement.
Concretely, as shown in
Next, a description will be given of the signal delay time measurement process.
As shown in
Next, in the sound space 260, the background noise is measured by the microphone 8 without outputting the measurement pulse signal (test signal), and the value is prescribed as the noise level Na (step S202). Subsequently, three counters, i.e., Counter_a, Counter_b and Burst, are cleared (i.e., counter value=0) (step S203). The Counter_a shows the total number of times of measurement. The Counter_b shows which measurement of the initial set number, the first additional number and the second additional number the current measurement is included in. Concretely, if Counter_b=0, the current measurement is the measurement during the initial set number. If Counter_b=1, the current measurement is the measurement during the first additional number. If Counter_b=2, the current measurement is the measurement during the second additional set number.
Next, the synchronized addition data buffer 231 is cleared (step S204). Then, the sound field measurement process is executed (step S205).
In this manner, the first measurement is executed. Specifically, first, the microphone 8 starts capturing the sound in the sound space 260, and the measurement pulse signal is outputted as the test signal (step S303). Thereby, the response signal by the first measurement is obtained and stored in the microphone input buffer 232.
Next, it is determined whether or not Counter_a=0 (step S304). Since Counter_a=0 in the first measurement, the process goes to step S306. Then, from the response signal stored in the microphone input buffer 232, the noise level Nb in the in-device delay time Tp is calculated (step S306). As described above, the noise level Nb shows the noise level in no-response period in which no response component of the sound space to the measurement pulse signal arrives.
Next, the response signal in the microphone input buffer 232 is supplied to the synchronized addition data buffer 231, and the response data after the synchronized addition is stored (step S307). Then, Counter_a and Counter_c are incremented, respectively (steps S308 and S309).
Next, it is determined whether or not Counter_c becomes equal to or larger than the variable P (step S310). Thereby, it is determined whether or not the measurements of the initial set number (four times in this embodiment) end. In such a case that step S310 is No, the process goes back to step S303, and steps S303 to S310 are repeated. In this manner, when the measurements of the initial set number end (step S310; Yes), Counter_b is incremented (step S311), and the process goes back to the sound field determination process shown in
When it is determined that the value of Counter_a is not 0 in step S304, i.e., in a case of the second or subsequent measurement, the above-mentioned correlation determination is executed with using the past response signal data (step S305). Then, when it is determined that the correlation between the response signal obtained by this measurement and the past response signal data is smaller than the predetermined reference, “1” is set to a flag Burst. The flag Burst is the flag showing the presence or absence of the above-mentioned unexpected noise. When the unexpected noise is detected, “1” is set to the flag Burst.
Returning to the sound field determination process shown in
Next, it is determined whether or not Counter_b=2 (step S208). As described above, Counter_b shows which state of the initial set number, the first additional number and the second additional number the current measurement is included in. “Counter_b=2” means that all of the initial set number, the first additional number and the second additional number are completed. Therefore, when step S208 is Yes, the sound field determination process ends.
Meanwhile, when step S208 is No, it is determined whether or not the flag Burst=1 (step S209). That step S209 is Yes means that the unexpected noise is detected in the past measurement. Thus, in order to remove the influence of the unexpected noise, the process goes back to step S205, and the sound field measurement process is repeated.
When step S209 is No, the S/N is calculated with using the noise level N obtained in step S206 and the signal level S obtained in step S207, and it is determined whether or not the S/N is larger than the smallest value SNref of the desired S/N (step S210). When the S/N is larger than the desired S/N value, since the response signal data obtained by the past measurement satisfies the desired S/N value, the process goes back to the signal delay time measurement shown in
In this manner, the sound field measurement process is repeatedly executed until the desired S/N is obtained (step S210; Yes) or until the measurements in all of the initial set number, the first additional number and the second additional number are completed. As a result, the desired S/N is obtained by the effect of the synchronized addition of the response signal data in the plural measurements, or based on the response signal data obtained after execution of the measurements of the maximum number, the subsequent signal delay time measurement is executed. In addition, when the unexpected noise is detected during the measurement, the measurement is further repeated in order to remove the influence. Hence, in any case, it becomes possible to obtain the response signal data with high accuracy in the minimum of time.
When the sound field determination process ends in this manner, the process returns to the signal delay time measurement process shown in
Next, a description will be given of the measurement method of the noise level. In the above embodiment, the noise level Na is measured before the execution of the sound field determination process (step S202, hereinafter also referred to as “pre-measurement”), and the noise level Nb in the in-device delay time Tp is measured in each sound field process (step S306, hereinafter also referred to as “immediate measurement”). The largest value of the noise levels Na and Nb is prescribed as the noise level N, and the S/N is calculated. However, this is not necessary. Namely, only the pre-measurement or the immediate measurement may be employed.
When only the pre-measurement is employed, the processes of steps S206 and S306 may be omitted. When the variation of the noise level N is sufficiently small and it can be regarded that the S/N is not varied, only the pre-measurement may be executed. In this case, there is such advantage that, since the state of the noise is initially defined, the S/N can be obtained by measuring the signal level S only once and the number of times of measurement can be determined at the early stage.
On the other hand, when only the immediate measurement is employed, the processes of steps S202 and S206 may be omitted. As understood from the processes shown in
Next, the description will be given of an embodiment of the automatic sound field correcting system to which the present invention is applied, with reference to the attached drawings.
In
While the audio system 100 includes the multi-channel signal transmission paths, the respective channels are referred to as “FL-channel”, “FR-channel” and the like in the following description. In addition, the subscripts of the reference number are omitted to refer to all of the multiple channels when the signals or components are expressed. On the other hand, the subscript is put to the reference number when a particular channel or component is referred to. For example, the description “digital audio signals S” means the digital audio signals SFL to SSBR, and the description “digital audio signal SFL” means the digital audio signal of only the FL-channel.
Further, the audio system 100 includes D/A converters 4FL to 4SBR for converting the digital output signals DFL to DSBR of the respective channels processed by the signal processing by the signal processing circuit 2 into analog signals, and amplifiers 5FL to 5SBR for amplifying the respective analog audio signals outputted by the D/A converters 4FL to 4SBR. In this system, the analog audio signals SPFL to SPSBR after the amplification by the amplifiers 5FL to 5SBR are supplied to the multi-channel speakers 6FL to 6SBR positioned in a listening room 7, shown in
The audio system 100 also includes a microphone 8 for collecting reproduced sounds at a listening position RV, an amplifier 9 for amplifying a collected sound signal SM outputted from the microphone 8, and an A/D converter 10 for converting the output of the amplifier 9 into a digital collected sound data DM to supply it to the signal processing circuit 2.
The audio system 100 activates full-band type speakers 6FL, 6FR, 6C, 6RL, 6RR having frequency characteristics capable of reproducing sound for substantially all audible frequency bands, a speaker 6WF having frequency characteristics capable of reproducing only low-frequency sounds and surround speakers 6SBL and 6SBR positioned behind the listener, thereby creating sound field with presence around the listener at the listening position RV.
With respect to the positions of the speakers, as shown in
The signal processing circuit 2 may have a digital signal processor (DSP), and roughly includes a signal processing unit 20 and a coefficient operation unit 30 as shown in
The coefficient operation unit 30 receives the signal collected by the microphone 8 as the digital collected sound data DM, generates the coefficient signals SF1 to SF8, SG1 to SG8, SDL1 to SDL8 for the frequency characteristics correction, the level correction and the delay characteristics correction, and supplies them to the signal processing unit 20. The signal processing unit 20 appropriately performs the frequency characteristics correction, the level correction and the delay characteristics correction based on the collected sound data DM from the microphone 8, and the speakers 6 output optimum sounds.
As shown in
The frequency characteristics correction unit 11 adjusts the frequency characteristics of the equalizers EQ1 to EQ8 corresponding to the respective channels of the graphic equalizer GEQ. The inter-channel level correction unit 12 controls the attenuation factors of the inter-channel attenuators ATG1 to ATG8, and the delay characteristics correction unit 13 controls the delay times of the delay circuits DLY1 to DLY8. Thus, the sound field is appropriately corrected.
The equalizers EQ1 to EQ5, EQ7 and EQ8 of the respective channels are configured to perform the frequency characteristics correction for each frequency band. Namely, the audio frequency band is divided into 9 frequency bands (each of the center frequencies are f1 to f9), for example, and the coefficient of the equalizer EQ is determined for each frequency band to correct frequency characteristics. It is noted that the equalizer EQ6 is configured to control the frequency characteristics of low-frequency band.
The audio system 100 has two operation modes, i.e., an automatic sound field correcting mode and a sound source signal reproducing mode. The automatic sound field correcting mode is an adjustment mode, performed prior to the signal reproduction from the sound source 1, wherein the automatic sound field correction is performed for the environment that the audio system 100 is placed. Thereafter, the sound signal from the sound source 1 such as a CD player is reproduced in the sound source signal reproduction mode. An explanation below mainly relates to the correction operation in the automatic sound field correcting mode.
With reference to
The switch elements SW11, SW12 and SWN are controlled by the system controller MPU configured by microprocessor shown in
The inter-channel attenuator ATG1 is connected to the output terminal of the equalizer EQ1, and the delay circuit DLY1 is connected to the output terminal of the inter-channel attenuator ATG1. The output DFL of the delay circuit DLY1 is supplied to the D/A converter 4FL shown in
The other channels are configured in the same manner, and switch elements SW21 to SW81 corresponding to the switch element SW11 and the switch elements SW22 to SW82 corresponding to the switch element SW12 are provided. In addition, the equalizers EQ2 to EQ8, the inter-channel attenuators ATG2 to ATG8 and the delay circuits DLY2 to DLY8 are provided, and the outputs DFR to DSBR from the delay circuits DLY2 to DLY8 are supplied to the D/A converters 4FR to 4SBR, respectively, shown in
Further, the inter-channel attenuators ATG1 to ATG8 vary the attenuation factors within the range equal to or smaller than 0 dB in accordance with the adjustment signals SG1 to SG8 supplied from the inter-channel level correction unit 12. The delay circuits DLY1 to DLY8 control the delay times of the input signal in accordance with the adjustment signals SDL1 to SDL8 from the phase characteristics correction unit 13.
The frequency characteristics correction unit 11 has a function to adjust the frequency characteristics of each channel to have a desired characteristic. As shown in
The band-pass filter 11a is configured by a plurality of narrow-band digital filters passing 9 frequency bands set to the equalizers EQ1 to EQ8. The band-pass filter 11a discriminates 9 frequency bands each including center frequency f1 to f9 from the collected sound data DM from the A/D converter 10, and supplies the data [PxJ] indicating the level of each frequency band to the gain operation unit 11c. The frequency discriminating characteristics of the band-pass filter 11a is determined based on the filter coefficient data stored, in advance, in the coefficient table 11b.
The gain operation unit 11c operates the gains of the equalizers EQ1 to EQ8 for the respective frequency bands at the time of the automatic sound field correction based on the data [PxJ] indicating the level of each frequency band, and supplies the gain data [GxJ] thus operated to the coefficient determination unit 11d. Namely, the gain operation unit 11c applies the data [PxJ] to the transfer functions of the equalizers EQ1 to EQ8 known in advance to calculate the gains of the equalizers EQ1 to EQ8 for the respective frequency bands in the reverse manner.
The coefficient determination unit 11d generates the filter coefficient adjustment signals SF1 to SF8, used to adjust the frequency characteristics of the equalizers EQ1 to EQ8, under the control of the system controller MPU shown in
In other words, the coefficient table 11e stores the filter coefficient data for adjusting the frequency characteristics of the equalizers EQ1 to EQ8, in advance, in a form of a look-up table. The coefficient determination unit 11d reads out the filter coefficient data corresponding to the gain data [GxJ], and supplies the filter coefficient data thus read out to the respective equalizers EQ1 to EQ8 as the filter coefficient adjustment signals SF1 to SF8. Thus, the frequency characteristics are controlled for the respective channels.
Next, the description will be given of the inter-channel level correction unit 12. The inter-channel level correction unit 12 has a role to adjust the sound pressure levels of the sound signals of the respective channels to be equal. Specifically, the inter-channel level correction unit 12 receives the collected sound data DM obtained when the respective speakers 6FL to 6SBR are individually activated by the measurement signal (pink noise) DN outputted from the measurement signal generator 3, and measures the levels of the reproduced sounds from the respective speakers at the listening position RV based on the collected sound data DM.
The level detection unit 12a detects the level of the collected sound data DM, and carries out gain control so that the output audio signal levels for all channels become equal to each other. Specifically, the level detection unit 12a generates the level adjustment amount indicating the difference between the level of the collected sound data thus detected and a reference level, and supplies it to an adjustment amount determination unit 12b. The adjustment amount determination unit 12b generates the gain adjustment signals SG1 to SG8 corresponding to the level adjustment amount received from the level detection unit 12a, and supplies the gain adjustment signals SG1 to SG8 to the respective inter-channel attenuators ATG1 to ATG8. The inter-channel attenuators ATG1 to ATG8 adjust the attenuation factors of the audio signals of the respective channels in accordance with the gain adjustment signals SG1 to SG8. By adjusting the attenuation factors of the inter-channel level correction unit 12, the level adjustment (gain adjustment) for the respective channels is performed so that the output audio signal level of the respective channels become equal to each other.
The delay characteristics correction unit 13 adjusts the signal delay resulting from the difference in distance between the positions of the respective speakers and the listening position RV. Namely, the delay characteristics correction unit 13 has a role to prevent that the output signals from the speakers 6 to be listened simultaneously by the listener reach the listening position RV at different times. Therefore, the delay characteristics correction unit 13 measures the delay characteristics of the respective channels based on the collected sound data DM which is obtained when the speakers 6 are individually activated by the measurement signal DN outputted from the measurement signal generator 3, and corrects the phase characteristics of the sound field space based on the measurement result.
Specifically, by turning over the switches SW11 to SW82 shown in
In the present invention, the delay amount operation unit 13a includes each component shown in
Next, the description will be given of the operation of the automatic sound field correction by the automatic sound field correcting system employing the configuration described above.
First, as the environment in which the audio system 100 is used, the listener positions the multiple speakers 6FL to 6SBR in a listening room 7 as shown in
Next, the basic principle of the automatic sound field correction according to the present invention will be described. As described above, the processes executed in the automatic sound field correction are the frequency characteristics correction of each channel, the correction of the sound pressure level and the delay characteristics correction. The description will schematically be given of the automatic sound field correction process with reference to a flow chart shown in
First, in step S10, the frequency characteristics correction unit 11 adjusts the frequency characteristics of the equalizers EQ1 to EQ8. Next, in an inter-channel level correction process in step S20, the inter-channel level correction unit 12 adjusts the attenuation factors of the inter-channel attenuators ATG 1 to ATG 8 provided for the respective channels. Next, in a delay characteristics correction process in step S30, the delay characteristics correction unit 13 adjusts the delay time of the delay circuits DLY1 to DLY8 of all the channels. The automatic sound field correction according to the present invention is performed in this order.
Next, the operation for each process will be explained in order. First, the frequency characteristics correction process in step S10 will be explained with reference to
In
Next, the frequency characteristics correction is executed for each channel. Namely, the signal processing circuit 2 outputs the frequency characteristics measurement signal such as the pink noise for one channel, and the signal is outputted from the speaker 6 as the measurement signal sound (step S108). The measurement signal sound is collected by the microphone 8, and the collected sound data is obtained in the frequency characteristics correction unit 11 in the signal processing circuit 2 (step S110). The gain operation unit 11c in the frequency characteristics correction unit 11 analyzes the collected sound data, and the coefficient determination unit 11d sets the equalizer coefficient (step S112). On the basis of the equalizer coefficient, the equalizer is adjusted (step S114). Thereby, based on the collected sound data, the frequency characteristics correction is completed for one channel. The process is executed for all the channels (step S116; Yes), and the frequency characteristics correction process is completed.
Next, an inter-channel level correction process in step S20 is performed. The inter-channel level correction process is performed in accordance with the flow chart shown in
In the signal processing unit 20 shown in
Next, the delay characteristics correction process in step S30 is executed in accordance with a flow chart shown in
As described above, the delay amount operation unit 13a includes each component shown in
When the in-device delay time Tp passes (step S136; Yes), the no-response period ends. Therefore, the threshold determination unit 254 determines the threshold (step S138). The comparator 252 executes the comparison processing and calculates the signal delay amount Td (step S140).
The process is executed for all the other channels. When the process is completed for all the channels (step S142; Yes), the memory 13c stores the delay amount of all the channels. Next, based on storage contents of the memory 13c, the coefficient operation unit 13b determines the coefficients of the delay circuits DLY1 to DLY8 of the respective channels so that the signals of all the other channels simultaneously reach the listening position RV with respect to the channel having the largest delay amount in all the channels, and supplies them to the respective delay circuits DLYs (step S138). Thereby, the delay characteristics correction is completed.
In that way, the frequency characteristic, the inter-channel level and the delay characteristics are corrected, and the automatic sound field correction is completed.
In the above-mentioned embodiment, the signal process according to the present invention is realized by the signal processing circuit. Instead, if the identical signal process is designed as a program to be executed on a computer, the signal process can be realized on the computer. In that case, the program is supplied by a recording medium, such as a CD-ROM and a DVD, or by communication by using a network and the like. As the computer, a personal computer and the like can be used, and an audio interface corresponding to plural channels, plural speakers and microphones and the like are connected to the computer as peripheral devices. By executing the above-mentioned program on the personal computer, the measurement signal is generated by using the sound source provided inside or outside the personal computer, and is outputted via the audio interface and the speaker to be collected by using the microphone. Thereby, the above-mentioned sound characteristics measuring device and automatic sound field correcting device can be realized by using the computer.
Additionally, in the above embodiment, the characteristics measurement device according to the present invention is applied to the automatic sound field correction device for measuring the sound field characteristics. However, the characteristics measurement device according to the present invention is applicable to various kinds of characteristics measurements. For example, the characteristics measurement device is applicable to general distance measurements such as a light transmission characteristics, a wave transmission characteristics, an electric circuit characteristics and an inter-vehicular distance in a certain environment. As for the sound characteristics, the characteristics measurement device is applicable to a distance measurement, a level measurement, a frequency characteristics measurement, a standing wave measurement, a speaker large/small determination measurement and a speaker existence/absence determination measurement. Namely, the characteristics measurement device of the present invention is applicable to various kinds of measurement devices for measuring characteristics subjected to the measurement by outputting the test signal and measuring the response.
The present invention is applicable to a sound field control system used in an environment for reproducing sounds with using plural speakers.
| Number | Date | Country | Kind |
|---|---|---|---|
| 2005-054526 | Feb 2005 | JP | national |
| Filing Document | Filing Date | Country | Kind | 371c Date |
|---|---|---|---|---|
| PCT/JP2006/003761 | 2/28/2006 | WO | 00 | 10/31/2007 |
