JP2006267534A - Sound system - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a sound system which can conduct sound characteristic control that is suitable for the change in the environment. <P>SOLUTION: The sound system has a sound signal source means 1 which outputs sound signals, a signal processing means 2 which processes the sound signals output from the sound signal source means 1 and outputs a plurality of processed sound signals, amplifying means 4 to 7 which amplify the plurality of the processed sound signals, a plurality of speakers 8 to 11 which output respective sounds based on the plurality of the amplified and processed sound signals, and a sound collecting means 12 which collects the sounds output from the plurality of the speakers 8 to 11. The signal processing means 2 determines the process contents of the sound signals so as to optimize the sound characteristic at the listening position on the basis of the collected sounds by the sound collecting means 12. <P>COPYRIGHT: (C)2007,JPO&INPIT

Description

  The present invention relates to an acoustic system configured by, for example, connecting a plurality of acoustic devices to an in-vehicle network, and more particularly to an acoustic system that can be adjusted to optimize acoustic characteristics at a listening position.

  As an in-vehicle acoustic system, signal processing that takes into account the sound field in the passenger compartment is performed on the audio signal reproduced by an audio device (for example, a CD player), and audio based on the audio signal subjected to signal processing is output from each speaker. There is something to make. However, in general, the signal processing applied to the audio signal does not adapt in detail to the cabin environment for each user, and is processing based on data selected from data patterns prepared in advance, Alternatively, the listener himself or herself manually adjusts the acoustic characteristics.

  Moreover, according to the user's request, the acoustic characteristics in the vehicle interior can be transmitted to an external data distribution center, and the data distributed from the data distribution center can be used to flexibly cope with the vehicle cabin environment for each user. There is also a system proposal (for example, see Patent Document 1).

  Furthermore, in a system constructed on the basis of an in-vehicle network (optical network) compliant with the MOST (Media Oriented Systems Transport) standard, which is a European standard for in-vehicle networks, a document proposed for operating methods of connected A / V equipment ( However, there is no disclosure regarding control of acoustic characteristics based on an in-vehicle network.

JP 2003-91290 A JP 2001-223718 A

  However, the conventional acoustic system has a method of processing contents based on a data pattern prepared in advance or performing a manual adjustment of acoustic characteristics by the listener when performing signal processing on an audio signal. Adopted, it did not perform signal processing that adapts in detail to the cabin environment for each user. Also, the method using data distributed from the data distribution center takes time because it requires communication with the outside, and there is also a problem that the apparatus becomes large and expensive due to equipment for communication with the outside. .

  Accordingly, the present invention has been made to solve the above-described problems of the prior art, and an object of the present invention is to provide an acoustic system capable of performing acoustic characteristic control adapted to environmental differences. is there.

  The acoustic system of the present invention includes an audio signal source unit that outputs an audio signal, a signal processing unit that processes the audio signal and outputs a plurality of processed audio signals, and each of the plurality of processed audio signals. A plurality of speaker means for outputting sound based on the sound collecting means for collecting sound output from the plurality of speaker means, and the signal processing means based on the sound collected by the sound collecting means, It is characterized in that the processing content of the audio signal is determined.

  According to the acoustic system of the present invention, since the processing content of the audio signal by the signal processing unit is determined based on the sound collected by the audio collecting unit, it is possible to perform acoustic characteristic control adapted to the difference in environment. There is an effect.

Embodiment 1 FIG.
FIG. 1 is a block diagram showing a configuration of an acoustic system according to Embodiment 1 of the present invention. As shown in FIG. 1, the acoustic system according to Embodiment 1 includes an audio signal source unit 1, a signal processing unit 2, a transmission medium 3, signal amplification units 4 to 7, and signal amplification units 4 to 7. Speakers 8 to 11 connected to each other, voice collecting means 12, and a microphone 13 connected to the voice collecting means 12.

  The audio signal source means 1 is a playback device such as a CD player, for example, and outputs an audio signal in a digital signal format. The signal processing unit 2 performs signal processing on the audio signal output from the audio signal source unit 1 and outputs a plurality of processed audio signals to a plurality of systems. The signal amplifying means 4 to 7 receive and amplify signals assigned to themselves among the plurality of processed audio signals output from the signal processing means 2, and send the amplified signals to the speakers 8 to 11. Output.

  The sound collection means 12 outputs the sound signal output from the microphone 13 based on the sound acquired by the microphone 13 as a sound signal in a digital signal format.

  The transmission medium 3 is, for example, an optical network that conforms to the MOST standard. The transmission medium 3 is connected to an audio signal source unit 1, a signal processing unit 2, signal amplification units 4 to 7, and an audio collection unit 12. Various signals are transmitted through the transmission medium 3 from the audio signal source unit 1 to the signal processing unit 2, from the audio collection unit 12 to the signal processing unit 2, and from the signal processing unit 2 to the signal amplification units 4 to 7.

  FIG. 2 is a diagram illustrating the positions of the speakers 8 to 11 and the microphone 13 when the acoustic system according to Embodiment 1 is applied to the automobile 30 (that is, the in-vehicle acoustic system). In the example shown in FIG. 2, the speakers 8 and 9 are disposed in the front (front) 31 of the vehicle 30, the speakers 10 and 11 are disposed in the rear (rear) 32, and the rear left seat (the left of the rear seat 33). The microphones 13 are arranged in the column). The position of the microphone 13 indicates a measurement point of the acoustic characteristics, and an optimum listening environment is obtained in the rear left seat. As a method for correcting acoustic characteristics, there are a time alignment method for correcting adverse effects caused by a difference in sound arrival time caused by a difference in distance between the speakers 8 to 11 and the listening position, and a standing wave generated in the vehicle interior. An equalizer method for correcting non-flatness of the frequency characteristic is generally used. Therefore, the case of using the time alignment method and the equalizer method will be described as the function of the signal processing means 2 in the first embodiment. However, the present invention is not limited to these methods. Further, the positions and the number of the speakers 8 to 11 and the position and the number of the microphones 13 are not limited to the illustrated example.

  In the case of FIG. 2, the distances between the speakers 8 to 11 in the vehicle interior and the microphones 13 corresponding to the listening positions are different from each other. Therefore, the time until the sounds output from the speakers 8 to 11 reach the microphones 13 ( Arrival time) is different. In the time alignment method, an audio signal of a channel having a fast sound arrival time at the listening position is delayed, and correction is performed so that the sound arrival times from the speakers 8 to 11 are equalized at the listening position. For example, in terms of the distance relationship in FIG. 2, the sound reaches the microphone 13 in the order of the sound from the speaker 11, the sound from the speaker 10, the sound from the speaker 9, and the sound from the speaker 8. The audio signals input to the speaker 11, the speaker 10, and the speaker 9 are delayed so as to match the time.

FIG. 3 is a block diagram showing a configuration of the signal processing means 2 of the acoustic system according to the first embodiment. As shown in FIG. 3, the signal processing means 2 (hereinafter referred to as "I / F circuit".) Interface circuits responsible for transferring signals to the transmission medium 3 2a and, generating a reference speech signal S _r for measurement a signal generating circuit 2b for, select circuit 2c for switching between the original audio signals S ₁ from the reference speech signal S _r and the audio signal source 1, the original speech on the basis of the microphone sound signal S _m from the sound collecting means 12 and a processing circuit 2d applying a process to the signal S _1. I / F circuit 2a outputs to the select circuit 2c retrieves the original audio signals S ₁ to be transmitted from the audio signal source means 1 from the transmission medium 3, the microphone sound signal S _m to be transmitted from the sound collecting means 12 Output to the processing circuit 2d. Further, I / F circuit 2a, a 4 after treatment system speech signals _{S 4, S 5, S 6} , S 7 output from the processing circuit 2d, to the transmission medium 3 to correspond to the signal amplifying means 4-7 Is output.

  FIG. 4 is a block diagram showing the configuration of the audio signal source means 1, the signal amplifying means 4, and the audio collection means 12. As shown in FIG. 4, the audio signal source means 1 includes an audio signal source 1a and an I / F circuit 1b. The voice collecting unit 12 includes an A / D circuit 12a that converts an input signal from the microphone 13 into a digital signal format, and an I / F circuit 12b. Further, the signal amplifying means 4 includes an I / F circuit 4a and an amplifier circuit 4b. Each of the signal amplification means 5 to 7 has the same configuration as the signal amplification means 4. The I / F circuits 1 b, 4 b, 12 b in each means transfer signals to the transmission medium 3 in the same manner as the I / F circuit 2 a of the signal processing means 2.

FIGS. 5A to 5J are waveform diagrams for explaining the operation in the case where processing based on the time alignment method is performed in the signal processing means 2 of the acoustic system according to Embodiment 1. FIG. FIGS. 5A and 5F show measurement impulse signals (reference audio signal S _r ), and FIGS. 5B to 5E show microphone-collected sounds by outputs of the speakers 8 to 11 before time alignment processing. 5 (g) to 5 (j) show microphone collection sound signals output from the speakers 8 to 11 after the time alignment processing.

The procedure of the time alignment processing is to obtain the time (arrival time) and the arrival time difference until the sound output from the speakers 8 to 11 first reaches the microphone 13, and the path with the latest arrival time (each speaker 8 to 11). And a process of giving a delay to the audio signals corresponding to the speakers 8 to 11 so as to match the route from the microphone 13 to the microphone 13. During the operation for obtaining the arrival time difference, the signal generation circuit 2b generates an impulse signal (FIG. 5A) for a time alignment technique based on an instruction from the processing circuit 2d. At this time, the select circuit 2c selects the measurement impulse signal (reference speech signal) S _r, and output to the processing circuit 2d. Processing circuit 2d is processed sequentially outputs measurement impulse signal S _r corresponding to the speaker 8-11 with respect to the path of the audio signal, measures the arrival time of each speaker 8-11 from microphone sound collection signals, respectively To do. FIGS. 5B to 5E show the difference in arrival time measured before the time alignment process. Note that the waveforms shown in FIGS. 5A to 5J are drawn as similar impulse waveforms for the sake of simplicity, but in reality, the waveforms shown in FIG. The waveforms of b) to (e) and (g) to (j) are different. However, in general, since the direct wave arrives earliest, if the arrival time is determined at the leading rising edge of the measurement signal waveform, it is not affected by the reflected wave or the like. Therefore, the times Tb, Tc, Td, and Te in FIGS. 5A to 5J indicate the arrival times of the outputs of the speakers 8 to 11. The difference in arrival time of the outputs of the speakers 8 to 11, that is, the difference in arrival time corresponds to the shortest path (distance) from each speaker 8 to 11 to the microphone 13 as shown in FIG.

  Based on the arrival time difference obtained for each of the speakers 8 to 11 in this way, the processing circuit 2d gives a time delay to the audio signal corresponding to each of the speakers 8 to 11, and the time alignment shown in FIGS. The arrival time of the output of the speakers 8 to 11 is corrected like the microphone collected sound signal from the output of the speakers 8 to 11 after processing. When the processing content is confirmed, the selection circuit 2c selects the original audio signal from the audio signal source means 1 to the processing circuit 2d, and shifts to a normal operation with time alignment processing.

  FIGS. 6A to 6C are graphs for explaining the operation when the processing based on the equalizer technique is performed in the signal processing means 2 of the acoustic system according to Embodiment 1. FIG. The configuration of the signal processing means 2 that performs processing based on the equalizer technique is the same as that of the signal processing means 2 that performs processing based on the time alignment technique, and the operation contents of the signal generation circuit 2b and the processing circuit 2d are the same as the time alignment technique. It simply replaces the equalizer method.

  The processing procedure based on the equalizer method is to first obtain the frequency characteristics of the sounds from the speakers 8 to 11 individually by the octave bandpass filter, detect the peak or dip frequency and gain of each frequency characteristic, Correction is performed so that the characteristics are flattened. During the operation for obtaining the frequency characteristics by the speakers 8 to 11, the signal generation circuit 2a generates a pink noise signal for the equalizer method based on an instruction from the processing circuit 2d. As shown in FIG. 6A, the pink noise has a slope of −3 dB / oct (that is, the level decreases by 3 dB every time the frequency is doubled), and the higher the frequency, the higher the level. It has the characteristic that falls. When a pink noise signal is measured with an octave bandpass filter, as shown in FIG. 6B, the energy is uniform in any octave, so the pink noise signal is generally used as a measurement signal. Yes. FIG. 6C shows frequency characteristics of the microphone audio signal collected by the microphone 13 measured by the octave bandpass filter in the processing circuit 2d when pink noise is emitted from one of the speakers 8 to 11. . Non-flatness of the frequency characteristic due to the standing wave generated in the passenger compartment occurs, and the frequency characteristic is, for example, shown by a broken line. The frequency and gain of this peak or dip are detected, for example, an IIR (Infinite Impulse Response) type digital filter coefficient for flattening the peak or dip is obtained based on a correction table prepared in advance, and this filter processing is performed on the audio signal. And correct the frequency characteristics. The solid line in FIG. 6C is the corrected characteristic, and is corrected so that the frequency characteristic is flattened.

  FIG. 7 is a flowchart showing a processing procedure of the sound system according to the first embodiment. At the start of the operation, first, the user operates whether or not to enter the correction coefficient measurement mode (step ST1). When entering the measurement mode, the user performs an operation of measuring a correction coefficient for time alignment processing or measuring a correction coefficient for equalizer processing (step ST2).

  When measuring the correction coefficient of the time alignment process, the arrival times of the output sounds of the speakers 8 to 11 to the microphone 13 are sequentially measured (steps ST3 to ST6), and the respective delay times are determined (step ST7). In order to reduce the influence of erroneous detection, the measurement and determination in steps ST3 to ST7 are repeated a plurality of times (N times) and averaged (step ST8), and a time alignment correction coefficient is determined (step ST9).

  When measuring the correction coefficient of the equalizer process, the peak or dip frequency and gain of each frequency characteristic are measured for the output sounds of the speakers 8 to 11 (steps ST10 to ST13), and the frequency characteristics are analyzed (step ST14). . In order to reduce the influence of erroneous detection, the measurement and determination in steps ST10 to ST14 are repeated a plurality of times (N times) and averaged (step ST15), and an equalizer correction coefficient is determined (step ST16).

After completing the measurement mode in this way, the process proceeds to time alignment processing and / or equalizer processing (step ST1). As normal operation in the signal processing unit 2 with respect to the original audio signals S ₁ from the audio signal source means 1, the measurement result subjected to the time alignment processing and / or equalizer processing based on the (time alignment correction coefficients and equalizer correction coefficient) ( Steps ST17 and ST18), the processed audio signals S ₄ , S ₅ , S ₆ and S ₇ are output to the signal amplifying means 4 to 7.

Incidentally, in the above example, it provided with a signal generating circuit 2b to the signal processing unit 2, although the signal generating circuit 2b is described as a configuration for generating a reference speech signal S _r for measurement, signal a signal generating circuit 2b A configuration may be adopted in which the signal generating operation is controlled from the signal processing unit 2 by being provided outside the processing unit 2 (for example, in the audio signal source unit 1 and the signal amplification units 4 to 7).

  As described above, the acoustic system according to Embodiment 1 can perform acoustic characteristic control adapted to environmental differences. Moreover, when performing an equalizer process, the acoustic characteristic implement | achieved in each vehicle interior can be equalized. Furthermore, the acoustic characteristic adjustment can be automated by adopting a configuration in which the reference voice signal generation operation and the voice collection operation for voice collection are performed in cooperation.

  Further, the signal processing means 2 may be included in the audio signal source means 1 without making the processing function of the signal processing means 2 independent.

  FIG. 8 is a block diagram illustrating a configuration example in which a class D amplifier is used as the amplifier circuit 4b in the signal amplifying unit 4 of the acoustic system according to the first embodiment. The signal amplifying means 5 to 7 have the same configuration as the signal amplifying means 4. As shown in FIG. 8, the amplifier circuit 4b performs a noise shaping process on the audio signal from the I / F circuit 4a to convert it into a PWM signal (pulse width modulation signal), and based on the PWM signal. It has a power driver 4b2 that performs a switching operation, and an LC filter 4b3 that drives the speaker 8 with an analog audio signal by removing the PWM carrier. Class D amplifiers are becoming more and more popular due to their small size and high efficiency, and high sound quality output without crossover distortion. However, because the output stage is a switching operation, strong high-frequency noise is superimposed on the speaker drive line. There's a problem. Therefore, by arranging the signal amplifying means 4 to 7 on the speaker side via the in-vehicle network as in this configuration, the speaker drive line can be minimized, and unnecessary radiation can be generated while taking advantage of the class D amplifier. Generation can be reduced.

Embodiment 2. FIG.
FIG. 9 is a block diagram showing a configuration of an acoustic system according to Embodiment 2 of the present invention. As shown in FIG. 9, the acoustic system according to the second embodiment includes an audio signal source unit 1, a signal processing unit 2, a transmission medium 3, signal amplification / audio collection processing units 14 to 17, and signal amplification. / Speakers 8 to 11 connected to the sound collection processing means 14 to 17 and microphones 18 to 21 connected to the signal amplification / sound collection processing means 14 to 17. The signal amplification / sound collection unit 14 includes an I / F circuit 14a, an amplification circuit 14b, and an A / D circuit 14c. The other signal amplification / sound collection means 15 to 17 have the same configuration as the signal amplification / sound collection means 14.

  FIG. 10 is a diagram illustrating the positions of the speakers 8 to 11 and the microphones 18 to 21 when the acoustic system according to Embodiment 2 is applied to the automobile 40 (that is, the in-vehicle acoustic system). In the example shown in FIG. 10, the speakers 8 and 9 and the microphones 18 and 19 are arranged in the front (front) 41 of the vehicle 40, and the speakers 10 and 11 and the microphones 20 and 21 are arranged in the rear (rear) 42. The rear left seat (the left row of the rear seat 42) is the listening position 22. FIG. 10 shows an example using two front and rear speakers and microphones. In the drawing, the microphones 18 to 21 are described inside the passenger compartment with respect to the speakers 8 to 11, respectively. However, the positional relationship is not limited to this. It is arranged at the four corners. The positions of the microphones 18 to 21 indicate the measurement points of the acoustic characteristics in the second embodiment, and are different from the first embodiment in that the listening position 22 of the rear left seat and the measurement point are different positions. . In other words, the second embodiment is intended to obtain an optimum listening environment at the listening position 22 that is a position different from the measurement point (microphone position). The method for correcting acoustic characteristics in the second embodiment is basically the same as that in the first embodiment.

  FIGS. 11-14 is explanatory drawing about the case where the correction | amendment by the time alignment method in the listening position 22 is performed in the acoustic system which concerns on Embodiment 2. FIGS. In FIGS. 11 to 14, for simplicity of explanation, it is assumed that the speakers 8 to 11 are in the same positions as the microphones 18 to 21, and only the microphones 18 to 21 are illustrated. The listening position 22 is represented by coordinates (X, Y) with the microphone 18 drawn at the upper left in FIGS.

FIG. 11 shows a case where sounds based on the measurement impulse signals are sequentially output from the speakers 8 to 11 and the arrival time is measured by the microphone 21. In FIG. 11, A ₂₁ is the shortest path (also referred to as distance or path length) from the speaker 8 to the microphone 21, B ₂₁ is the shortest path from the speaker 9 to the microphone 21, and C ₂₁ is the shortest path from the speaker 10 to the microphone 21. It is a route. Shortest path _{D 11} from the speaker 11 to the microphone 21 is not shown because the speaker 11 and the microphone 21 overlaps. In FIG. 11, A _a is the shortest path (also referred to as distance or path length) from the speaker 8 to the listening position 22, B _a is the shortest path from the speaker 9 to the listening position 22, and C _a is listening from the speaker 10. This is the shortest route to the position 22. The arrival times that are output from the speakers 8 to 11 and reach the microphone 21 through the paths A ₂₁ , B ₂₁ , C ₂₁ , and D ₂₁ are T _A21 , T _B21 , T _C21 , and T _D21 , respectively. Here, T _D21 is time to substantially the signal processing means for measuring an impulse signal S _r output from 2 is output as sound from the speakers 8 to 11, the sound is captured by the microphone 21 signal Since it can be regarded as the sum of the time required to reach the processing means 2, the actual sound arrival time is expressed by the following equations (1) to (3).
Arrival time of path _{A 21 = T A21 -T D21 (} 1)
Route B ₂₁ arrival time = T _B21 −T _D21 (2)
Arrival time of path _{C 21 = T C21 -T D21 (} 3)
Further, it is possible to determine the equation (1) to (3) and the sound velocity _{V s} each path length from the _{A 21, B 21, C 21} .

Furthermore, the path lengths A _a , B _a , and C _a from the speakers 8 to 11 to the listening position are expressed by the following equations (4) to (6).
A _a = √ (X ² + Y ² ) (4)
B _a = √ (X ² + (C ₂₁ −Y) ² ) (5)
C _a = √ ((B ₂₁ −X) ² + Y ² ) (6)

Therefore, the arrival times of the routes A _a , B _a , and C _{a at} the listening position are as shown in the following equations (7) to (9).
Path _{A a} arrival time _{= (A a / A 21)} * (T A21 -T D21) (7)
Route _Ba arrival time = (B _a / B ₂₁ ) * (T _B21 −T _D21 ) (8)
Path _{C a} time of arrival _{= (C a / C 21)} * (T C21 -T D21) (9)

FIG. 12 shows a case where sounds based on the measurement impulse signal are sequentially emitted from the speakers 8 to 11 and the arrival time is measured by the microphone 20. 12, A ₂₀ is the shortest path (also referred to as distance or path length) from the speaker 8 to the microphone 20, B ₂₀ is the shortest path from the speaker 9 to the microphone 20, and D ₂₀ is the shortest path from the speaker 11 to the microphone 20. It is a route. Shortest path from the speaker 10 to the microphone 20 _{C 20} is not shown because the speaker 10 and the microphone 20 overlaps. In FIG. 12, A _a is the shortest path (also referred to as distance or path length) from the speaker 8 to the listening position 22, B _a is the shortest path from the speaker 9 to the listening position 22, and D _a is listening from the speaker 11. This is the shortest route to the position 22. The arrival times that are output from the speakers 8 to 11 and reach the microphone 20 through the paths A ₂₀ , B ₂₀ , and D ₂₀ are _denoted as T _A20 , T _B20 , T _C20 , and T _D20 , respectively. Here, _TD20 is the signal processing means that the time until the impulse signal for measurement substantially outputted from the signal processing means 2 is outputted as a sound from each of the speakers 8 to 11 and the sound is captured by the microphone 21. Since it can be regarded as the sum of the time required to reach 2, the actual sound arrival time is expressed by the following equations (10) to (12).
Arrival time of path _{A 20 = T A20 -T C20 (} 10)
Route B ₂₀ arrival time = T _B20 -TC ₂₀ (11)
Arrival time of path _{D 20 = T D20 -T C20 (} 12)
Further, it is possible to determine the equation (1) to (3) and the sound velocity _{V s} each path length from the _{A 20, B 20, D 20} .

Furthermore, the path lengths A _a , B _a , and D _a from the speakers 8 to 11 to the listening position are expressed by the following equations (13) to (15).
A _a = √ (X ² + Y ² ) (13)
B _a = √ (X ² + (D ₂₀ −Y) ² ) (14)
D _a = √ ((A ₂₀ −X) ² + (D ₂₀ −Y) ² ) (15)

Therefore, the arrival times of the respective routes A _a , B _a , and D _{a at} the listening position are expressed by the following equations (16) to (18).
Path _{A a} arrival time _{= (A a / A 20)} * (T A20 -T C20) (16)
Route _Ba arrival time = (B _a / B ₂₀ ) * (T _B20 -TC ₂₀ ) (17)
Path _{D a} arrival time _{= (D a / D 20)} * (T D20 -T C20) (18)

FIG. 13 shows a case where sounds based on the measurement impulse signal are sequentially emitted from the speakers 8 to 11 and the arrival time is measured by the microphone 19. In FIG. 13, A ₁₉ is the shortest path (also referred to as distance or path length) from the speaker 8 to the microphone 19, C ₁₉ is the shortest path from the speaker 10 to the microphone 19, and D ₁₉ is the shortest path from the speaker 11 to the microphone 19. It is a route. The shortest path B ₁₉ from the speaker 9 to the microphone 19 is not shown because the speaker 9 and the microphone 19 overlap. Further, in FIG. 13, A _a is (also referred to as distance or path length.) Shortest path to the listening position from the speaker 8, B _a is the shortest path to the listening position from the speaker 9, D _a distance from the listening position from the speaker 11 Is the shortest path. The arrival times that are output from the speakers 8 to 11 and reach the microphone 20 through the paths A ₁₉ , B ₁₉ , C ₁₉ , and D ₁₉ are T _A19 , T _B19 , T _C19 , and T _D19 , respectively. Here, _TD19 is the signal processing means that the time until the impulse signal for measurement substantially outputted from the signal processing means 2 is outputted as sound from each of the speakers 8 to 11 and the sound is captured by the microphone 21. Since it can be regarded as the sum of the time required to reach 2, the actual sound arrival time is expressed by the following equations (19) to (21).
Arrival time of path _{A 19 = T A19 -T B19 (} 19)
Arrival time of path _{C 19 = T C19 -T B19 (} 20)
Arrival time of path _{D 19 = T D19 -T B19 (} 21)
Further, it is possible to obtain the equation (19) to (21) and the acoustic velocity _{V s} each path length from the _{A 19,} C _{19, D 19.}

Accordingly, the arrival times of the respective routes A _a , C _a , and D _{a at} the listening position are expressed by the following equations (22) to (24).
A _a = √ (X ² + Y ² ) (22)
C _a = √ ((D ₁₉ −X) ² + Y ² ) (23)
D _a = √ ((D ₁₉ −X) ² + (A ₁₉ −Y) ² ) (24)

Accordingly, the arrival times of the respective routes A _a , C _a , and D _{a at} the listening position are expressed by the following equations (25) to (27).
Path _{A a} arrival time _{= (A a / A 19)} * (T A19 -T B19) (25)
Path _{C a} time of arrival _{= (C a / C 19)} * (T C19 -T B19) (26)
Path _{D a} arrival time _{= (D a / D 19)} * (T D19 -T B19) (27)

FIG. 14 shows a case where sounds based on the measurement impulse signal are sequentially emitted from the speakers 8 to 11 and the arrival time is measured by the microphone 18. In FIG. 14, B ₁₈ is the shortest path (also referred to as distance or path length) from the speaker 9 to the microphone 18, C ₁₈ is the shortest path from the speaker 10 to the microphone 18, and D ₁₈ is the shortest path from the speaker 11 to the microphone 18. It is a route. The shortest path B ₁₈ from the speaker 8 to the microphone 18 is not shown because the speaker 8 and the microphone 18 overlap each other. In FIG. 14, B _a is the shortest path from the speaker 9 to the listening position 22 (also referred to as distance or path length), C _a is the shortest path from the speaker 10 to the listening position 22, and D _a is listening from the speaker 11. This is the shortest route to the position 22. The arrival times that are output from the speakers 8 to 11 and reach the microphone 18 through the paths A ₁₈ , B ₁₈ , C ₁₈ , and D ₁₈ are _denoted as T _A18 , T _B18 , T _C18 , and T _D18 , respectively. Here, TA ₁₈ is a signal processing means that is a time until a measurement impulse signal substantially outputted from the signal processing means 2 is outputted as a sound from each of the speakers 8 to 11, and the sound is captured by the microphone 21. Since it can be regarded as the sum of the time required to reach 2, the actual sound arrival time is expressed by the following equations (28) to (30).
Route B ₁₈ arrival time = T _B18 −T _A18 (28)
Arrival time of path _{C 18 = T C18 -T A18 (} 29)
Arrival time of route D ₁₈ = T _D18 -T _A18 (30)
Further, it is possible to obtain the equation (28) - (30) and the acoustic velocity _{V s} each path lengths from _{B 18, C 18, D 18} .

Accordingly, the arrival times of the respective routes A _a , C _a , and D _{a at} the listening position are expressed by the following equations (31) to (33).
B _a = √ (X ² + (B ₁₈ −Y) ² ) (31)
C _a = √ ((C ₁₈ −X) ² + Y ² ) (32)
D _a = √ ((C ₁₈ −X) ² + (B ₁₈ −Y) ² ) (33)

Therefore, the arrival times of the respective paths B _a , C _a , and D _{a at} the listening position are expressed by the following equations (34) to (36).
Route _Ba arrival time = (B _a / B ₁₈ ) * (T _B18 −T _A18 ) (34)
Path _{C a} time of arrival _{= (C a / C 18)} * (T C18 -T A18) (35)
Path _{D a} arrival time _{= (D a / D 18)} * (T D18 -T A18) (36)

As described above, according to the measurements shown in FIGS. 11 to 14, for example, the arrival time of the route A _a from the speaker 8 to the listening position 22 is expressed by the equations (7), (16), and (25). The arrival time from the speaker 8 at the listening position 22 can be determined by averaging the obtained values. Others can be similarly determined. The time alignment correction based on these is the same as in the first embodiment.

  Similarly, in the equalizer processing, voices are individually collected in the microphones 18 to 21 based on the pink noise emitted from the speakers 8 to 11 in FIG. 10, and the frequency characteristics measured by the octave bandpass filter are obtained. As a result, the frequency characteristic at the measurement point of each microphone is obtained, and the filter coefficient for correcting the non-flatness of the frequency characteristic at each point can be determined. For the listening position 22, for example, the correction contents of the close measurement points are highly weighted (for example, weighted in inverse proportion to the path length), and the correction contents of the respective measurement points are synthesized. Applicable.

It is a block diagram which shows the structure of the acoustic system which concerns on Embodiment 1 of this invention. It is a figure which shows arrangement | positioning of a structure at the time of applying the acoustic system which concerns on Embodiment 1 to a motor vehicle. 3 is a block diagram illustrating a configuration of signal processing means of the acoustic system according to Embodiment 1. FIG. It is a block diagram which shows the structure of the audio | voice signal source means of a sound system which concerns on Embodiment 1, a signal amplification means, and an audio | voice collection means. (A)-(j) is a signal waveform diagram for demonstrating operation | movement in the case of performing a time alignment process in the signal processing means of the acoustic system which concerns on Embodiment 1. FIG. (A)-(c) is a graph for demonstrating operation | movement in the case of performing an equalizer process in the signal processing means of the acoustic system which concerns on Embodiment 1. FIG. 3 is a flowchart showing the operation of the acoustic system according to Embodiment 1. FIG. 3 is a block diagram showing a configuration using a class D amplifier as an amplifier circuit in the signal amplifying unit of the acoustic system according to the first embodiment. It is a block diagram which shows the structure of the acoustic system which concerns on Embodiment 2 of this invention. It is a figure which shows arrangement | positioning of a structure at the time of applying the acoustic system which concerns on Embodiment 2 to a motor vehicle. FIG. 12 is an operation explanatory diagram (part 1) when performing time alignment processing in the signal processing means of the acoustic system according to Embodiment 2. FIG. 12 is an operation explanatory diagram (part 2) when the time alignment process is performed in the signal processing unit of the sound system according to the second embodiment. FIG. 12 is an operation explanatory diagram (part 3) when the time alignment process is performed in the signal processing unit of the sound system according to the second embodiment. FIG. 11 is an operation explanatory diagram (part 4) when performing time alignment processing in the signal processing means of the acoustic system according to Embodiment 2.

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 Audio | voice signal source means, 2 Signal processing means, 3 Transmission medium, 4-7 Signal amplification means, 8-11 Speaker, 12 Voice collection means, 13 Microphone, 14-17 Signal amplification / voice collection means, 18-21 Microphone, 22 Listening position.

Claims (11)

Audio signal source means for outputting an audio signal;
Signal processing means for processing the audio signal and outputting a plurality of processed audio signals;
A plurality of speaker means for outputting a sound based on each of the plurality of processed audio signals;
Voice collecting means for collecting sounds output from the plurality of speaker means;
The acoustic system, wherein the signal processing means determines the processing content of the audio signal based on the sound collected by the sound collecting means. The signal processing means includes signal generation means for generating a reference audio signal,
The sound system according to claim 1, wherein the sound collected by the sound collecting means is a sound output from the plurality of speaker means based on the reference sound signal. A reference audio signal is input to the signal processing means,
The sound system according to claim 1, wherein the sound collected by the sound collecting means is a sound output from the plurality of speaker means based on the reference sound signal. The sound collected by the sound collecting means is a sound at a listening position,
The acoustic system according to any one of claims 1 to 3, wherein the signal processing means optimizes acoustic characteristics at the listening position based on the sound collected by the voice collecting means. A plurality of the voice collecting means;
The acoustic system according to any one of claims 1 to 3, wherein the signal processing unit optimizes acoustic characteristics at a listening position based on sounds collected by the plurality of voice collecting units. A plurality of signal amplifying means for amplifying a plurality of processed audio signals output from the signal processing means;
The sound system according to any one of claims 1 to 5, wherein the sound output from the plurality of speaker means is a sound based on the plurality of amplified post-processing audio signals. A transmission medium for transmitting the signal;
The acoustic system according to claim 6, wherein the audio signal source unit, the signal processing unit, the audio collection unit, and the signal amplification unit are connected to the transmission medium.   The acoustic system according to any one of claims 1 to 7, wherein the signal processing means is configured as a part of the audio signal source means.   9. The acoustic system according to claim 1, wherein the signal amplifying means includes a class D amplifier.   The acoustic system according to claim 1, wherein the acoustic system is a system mounted on an automobile.   The acoustic system according to claim 7, wherein the transmission medium is an in-vehicle network conforming to the MOST standard.

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