JP2006319786A - Sound field measuring apparatus and sound field measuring method - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To correct difference in characteristic between a plurality of microphones for sound field measurement. <P>SOLUTION: A sound field correction/measurement function section 31 in a sound field correction unit 3 is provided with a sound field correction processing block 32 for performing processing of correcting the characteristic of an original sound signal; and a measurement processing block 33 for measuring sound characteristic information required for analyzing a parameter to be provided to the original sound signal to generate a more realistic sound field. The measurement processing block 33 is further provided with measurement sections 331a, 331b, measurement sound processing sections 332a, 332b, and a microphone characteristic correction section 333. The measurement sound processing section 332 generates a measurement sound signal. The microphone characteristic correction section 333 corrects the sound collecting characteristic in microphones 6a, 6b and generates the extremely excellent sound field in its listening environment while adding this correction value. <P>COPYRIGHT: (C)2007,JPO&amp;INPIT

Description

  The present invention relates to a sound field measuring apparatus and a sound field measuring method that can generate or correct a predetermined sound field in real space by an audio signal output from a speaker, for example.

  In a playback device for video data, music data, etc., the sense of reality and sound quality are relatively easy for the user to judge. For example, when a user listens to an orchestra song, the position of each instrument can be clearly felt in the virtual sound field, and a sound field can be generated that is reminiscent of a real orchestra playing in front of you. Is preferred.

  For example, a two-channel stereo system that adjusts the balance of each signal channel of a two-channel stereo signal composed of an L signal and an R signal and outputs from two speakers so that the sound image of the reproduction sound field is localized at an optimal place as a virtual sound image, There are a three-channel stereo system in which a center speaker is added at the center of left and right two-channel speakers, and a 5.1 channel stereo system in which a rear speaker is further added.

  For example, in a multi-channel audio system such as a 5.1 channel stereo system, the parameters of the audio signals output from the speakers are determined so that the audio signals output from the speakers can reproduce a realistic sound field. However, for example, the balance and sound quality of the reproduced sound at the position where the listener listens change depending on the so-called listening environment including the structure of the listening room, the position of the listener relative to the speaker, etc. There has been a problem that the sound field (sound) that the listener actually feels may be different from the ideal reproduction sound field constructed during recording.

  This problem becomes prominent in small spaces such as small rooms and automobiles. Since the position of the listener is almost limited to the seat position in the interior of the automobile, the distance between the speaker and the listening position becomes large. Therefore, a time difference in the arrival time of the audio signal output from the speaker occurs, and the balance of the sound field is greatly disturbed. In particular, since the interior of an automobile is in a substantially sealed state, reflected sounds and the like are synthesized in a complex manner and reach the listener, which may be a factor that disturbs the reproduced sound field at the listening position. Furthermore, in small rooms and cars, the speaker mounting position is also limited, so when speaker placement in which the output sound from the speaker directly reaches the listener's ear cannot be realized, the change in sound quality due to speaker placement is also reproduced. It greatly affects the field degradation.

  For this reason, it is known that the listener performs appropriate acoustic correction on the output audio signal so that a reproduction sound field as close to the original sound field as possible is created according to the listening environment in which the audio system is used. ing. First, the acoustic characteristics in the listening environment are measured, and parameters for signal processing for performing acoustic correction on the audio output system of the audio set are set based on the measurement result. Then, by outputting an audio signal subjected to signal processing according to the parameters set here from the speaker, a good sound field corrected in conformity with the listening environment is reproduced. As this acoustic correction, for example, the frequency of the output sound may be corrected so that the sound signal to be output from each speaker has a delay time difference of zero at the listener's listening position (ear position).

  As an example of the measurement of the acoustic characteristics and the acoustic correction based on the measurement result, the following method using an acoustic correction device as disclosed in Patent Document 1 is known.

  First, a measurement microphone is arranged at a position (listening position) of a listener's ear in a space where an audio set is used, that is, a listening space. The measurement sound is output from the speaker by the acoustic correction device, and the measurement sound is collected by the microphone. From the characteristics of the collected sound signal, each speaker and the listening position (the microphone installation position, that is, the sound collection position) The distance information between is calculated. Since the arrival time of the sound in the space from each speaker to the listening position can be obtained based on the distance information, the acoustic correction device uses the information on the arrival time of each speaker to hear the sound emitted from each speaker. It is possible to set the delay time of the audio signal of the channel corresponding to each speaker so that the timings of arriving at are matched. This correction is generally referred to as time alignment.

JP 2000-261900 A

  When performing the above-described sound field measurement, if a single microphone is used, a parameter for performing a specific correction for the local state of the frequency of the reproduced audio signal in the listening environment or a change in frequency characteristics is selected. However, it is self-evident that a more flexible response can be achieved by performing the same measurement using a plurality of microphones and averaging the calculated values.

  For example, in the acoustic correction apparatus described above, a plurality of microphones are used to measure the sound field, but the characteristic difference between the microphones is an important problem. When the characteristic difference between the microphones is large, an inappropriate correction parameter may be calculated. For the purpose of measuring a correction parameter for giving an optimal correction for sound field reproduction, it is preferable that the difference in characteristics of each microphone is as small as possible. However, selecting a microphone with a small individual difference and incorporating it into an audio set or manufacturing the measurement microphone so that the quality difference between the individual microphones is small has led to an increase in manufacturing cost.

  Therefore, the present invention has been proposed in view of the above-described conventional situation, and provides a sound field measuring apparatus and a sound field measuring method capable of correcting a characteristic difference between a plurality of microphones for sound field measurement. The purpose is to do.

  In order to achieve the above-described object, a sound field measuring apparatus according to the present invention is a sound field that measures a sound field generated in a reproduction environment by an audio signal output from these speakers in an audio set including a plurality of speakers. In the measurement apparatus, a first microphone that collects audio signals output from a plurality of speakers installed at an arbitrary position in a reproduction environment, and an audio output from the plurality of speakers installed in the vicinity of the first microphone. A second microphone that collects the signal; a measurement unit that measures the sound collection characteristics of the audio signal in the first microphone and the sound collection characteristic of the audio signal in the second microphone; and the measured sound collection characteristics are analyzed. The characteristic analyzer and the voice signal collected by the first microphone and the second microphone are analyzed by the characteristic analyzer. A correction value generation unit that generates a correction value for performing correction to reduce the difference in sound collection characteristics between the first microphone and the second microphone based on the analysis value, and is collected by the first microphone and the second microphone; The sound field reproduced in the reproduction environment is measured more accurately by correcting the difference in the characteristics of the sound signal after it is sounded.

  When correcting the sound collection characteristics of the first microphone and the second microphone, one speaker among a plurality of speakers may be used as a reference. Since the first microphone and the second microphone are preferably installed at approximately the same distance from the reference speaker, in the characteristic analysis unit, the first microphone and the speaker to be measured among the plurality of speakers are connected. Analyzing the distance and the distance between the second microphone and the speaker to be measured, and when the distance between the two is substantially the same distance, the sound signal collected by the first microphone and the second microphone Correction is performed to reduce the sound collection characteristic difference between the first microphone and the second microphone.

  Examples of the sound collection characteristics include sound collection sensitivity in the first microphone or the second microphone, and frequency amplitude characteristics of the collected sound signal. The sound collection characteristic is calculated from an impulse response.

  In order to achieve the above-described object, the sound field measurement method according to the present invention measures a sound field generated in a reproduction environment by an audio signal output from these speakers in an audio set including a plurality of speakers. In the sound field measuring method, a first sound collecting step of collecting a sound signal output from a predetermined speaker among a plurality of sounds by a first microphone installed at an arbitrary position in a reproduction environment, and the first microphone A second sound collecting step of collecting a sound signal output from a predetermined speaker by a second microphone installed in the vicinity; a sound collecting characteristic of the sound signal acquired in the first sound collecting step; A measurement process for measuring the sound collection characteristics of the audio signal acquired in the sound collection process, a characteristic analysis process for analyzing the measured sound collection characteristics, and a first sound collection process and a second sound collection process. A correction value generation step for generating a correction value for performing a correction for reducing the difference in sound collection characteristics between the first microphone and the second microphone based on the analysis value analyzed in the characteristic analysis step with respect to the sound signal thus obtained; In the characteristic analysis step, the distance between the first microphone and the speaker that is the measurement target among the plurality of speakers and the distance between the second microphone and the speaker that is the measurement target are analyzed, and the distance between the two is approximately When the distance is equal, correction is performed to reduce the difference between the sound collection characteristics of the first microphone and the second microphone with respect to the sound signals collected by the first microphone and the second microphone.

  According to the present invention, it is possible to automatically correct a characteristic difference between a plurality of microphones installed in a reproduction environment in order to measure a sound field.

  Hereinafter, a sound field measuring apparatus shown as a specific example of the present invention will be described in detail with reference to the drawings. The sound field measuring apparatus 1 shown as a specific example includes a plurality of speakers and can reproduce the sound field at the time of recording more realistically by audio signals output from the speakers, and is an audio set corresponding to a so-called multi-channel method. To collect sound from a microphone as sound collection means for measurement to measure sound characteristics information required for analysis of parameters and the like given to the original sound signal in order to generate a more realistic sound field The characteristic can be corrected.

  FIG. 1 shows a configuration example of an entire audio set to which a sound field measuring apparatus according to the present invention is applied. An audio set 1 shown in FIG. 1 includes a media playback unit 2 that reads music content data recorded from a recording medium (hereinafter referred to as a medium), and a sound field that changes the characteristics of the played back multi-channel audio signal. A sound field correction unit 3 having a correction function and a function of correcting the sound collection characteristics of the microphone, a power amplifier unit 4 for amplifying each of the corrected multi-channel audio signals, and various speakers 51 to 5n, The apparatus includes two microphones 6a and 6b that measure a sound field generated by an audio signal output from a speaker. In addition, the audio set 1 includes a memory unit 8, a program for executing processing for correcting the sound field in the sound field correction unit 3, and processing for measuring the output signal from the speaker and correcting the sound collection characteristics of the microphone, Information necessary for the processing is stored. As the memory unit 8, for example, a nonvolatile and rewritable memory element such as a flash memory can be applied. Each of the above-described units is comprehensively controlled by the control unit 7.

  The media playback unit 2 reads audio content data recorded on the media. There are no particular restrictions on the type of media that can be played back by the media playback unit 2, the recording format, and the like, but examples include CD (Compact Disc) and DVD (Digital Versatile Disc).

  Current general DVD formats include DVD Audio, AC3 (Audio Code Number 3), etc., which conform to the DVD standard. In this specific example, audio data is compressed and encoded according to the AC3 method. Yes. Therefore, the media playback unit 2 also includes a decoder that decodes the compression-coded audio data.

  The media playback unit 2 may be a so-called combo drive that can play back both DVDs and audio CDs. Further, the input destination of the audio signal may be a television tuner that receives and demodulates a television broadcast or the like and outputs a video signal and an audio signal as well as a medium that can be reproduced by the media reproducing unit 2. Further, it may be a wired LAN, a wireless LAN, a network, or a large-scale network such as a so-called Internet in which such networks are connected to each other. Furthermore, it may be a large-capacity recording medium such as a hard disk. Furthermore, the media playback unit 2 may have the above-described configuration for media playback, a television tuner, a configuration for network connection, an HDD, and the like.

  When the media playback unit 2 supports multiple audio channels, the audio signal read out by the media playback unit 2 is output through a plurality of signal lines corresponding to each audio channel. In this specific example, the audio set 1 corresponds to the 5.1 channel surround system, and the media playback unit 2 has the maximum center channel (C), front left channel (FL), and front right channel (FR). 6 audio signals corresponding to the left surround channel (BL), the right surround channel (BR) and the subwoofer channel are output. The audio signal reproduced by the media reproducing unit 2 is input to the power amplifier unit 4 as a signal whose acoustic characteristics are corrected by the measurement function unit and the sound field correction function unit of the sound field correction unit 3. Details of the sound field correction unit 3 will be described later.

  The power amplifier unit 4 amplifies the input audio signal and outputs a drive signal for driving the speaker. Here, the power amplifier unit 4 includes a number of circuit systems corresponding to the channel configuration to which the audio set 1 corresponds, and amplifies the audio signal for each channel by each circuit, for example, in an appropriate position in the listening environment as described above. Drive signals are output to speakers corresponding to the center channel (C), front left channel (FL), front right channel (FR), left surround channel (BL), right surround channel (BR), and subwoofer channel. . With this multi-channel configuration, the audio set 1 can reproduce the recording environment when the music content is recorded in the current listening environment.

  The speaker 5 can be provided in a number corresponding to the number of channels. In this specific example, since the 5.1 channel surround system is used, a total of six speakers are connected for each channel. Further, if the 7.1 channel surround system is supported, eight speakers corresponding to each channel can be connected. The arrangement of speakers and microphones in the audio set 1 will be described with reference to FIG. FIG. 2 shows a typical speaker arrangement in an audio set corresponding to the 5.1 channel surround system.

  The speaker 51 shown in FIG. 2 is a center channel (C), the speaker 52 is a front left channel (FL), the speaker 53 is a front right channel (FR), the speaker 54 is a left surround channel (BL), and the speaker 55 is a right surround channel (FR). BR) respectively. In addition, the audio set 1 includes subwoofer channel speakers not shown in FIG. 2, and the media playback unit 2 outputs six audio signals corresponding to these six channels.

  A sound field is generated in a region surrounded by the speaker by the audio signal output from the speaker arranged as shown in FIG. Examples of the listening environment that uses the audio set 1 include the inside of a car and a small room.

  The microphones 6a and 6b are means for collecting a predetermined measurement sound when measuring the sound field generated in the listening environment. The microphones 6a and 6b are based on one speaker of a plurality of speakers. It is preferable that they are installed at approximately the same distance from the reference speaker. In this specific example, the microphone 6a and the microphone 6b are fixed to each other with an interval that does not cause a characteristic difference depending on the installation position of both in the listening environment. The sound signals collected by the microphones 6 a and 6 b are input to the sound field correction unit 3.

  The control unit 7 includes a microcomputer including a CPU (Central Processing Unit), a ROM, a RAM, and the like, and executes control and various processes for each unit constituting the audio set 1 shown in FIG. The control unit 7 may be connected to a user interface unit 9 for accepting an operation selection from the user.

  Next, the internal configuration of the sound field correction unit 3 will be described in detail with reference to FIG.

  The sound field correction unit 3 includes a sound field correction / measurement function unit 31 having a function of correcting a sound field and a function of measuring output sound of a speaker. The sound field correction / measurement function unit 31 includes a sound field correction processing block 32 that performs processing for correcting the characteristics of the original sound signal, and parameters that are given to the original sound signal in order to generate a more realistic sound field. And a measurement processing block 33 that measures voice characteristic information necessary for such analysis.

  The sound field correction / measurement function unit 31 includes a microphone amplifier 34a that amplifies the sound signal input from the microphone 6a and a microphone amplifier 34b that amplifies the sound signal input from the microphone 6b. The measurement signal amplified in 34b is sent to the measurement processing block 33 for measurement processing.

  The sound field correction processing block 32 performs a process for sound field correction based on the measurement result and changes a predetermined parameter value. The switch 35 is provided to switch between the measurement mode and the sound field correction mode. The switch 35 is switched so that the terminal Tm2 or Tm3 is alternatively connected to the terminal Tm1. The switching is controlled by the control unit 7.

  The measurement processing block 33 further includes measurement units 331a and 331b, measurement sound processing units 332a and 332b, and a microphone characteristic correction unit 333. The measurement sound processing unit 332 generates and outputs a measurement audio signal. Hereinafter, the sound signal for measurement is referred to as a measurement sound signal. The measurement sound signal is a special signal sound created by a CPU (Central Processing Unit) constituting the control unit 7 of the audio set 1 or a DSP (Digital Signal Processor) not shown. Therefore, the characteristic difference after sound collection with respect to the signal characteristic at the time of creating the measurement sound signal collected simultaneously by the microphones 6a and 6b can be analyzed by the DSP or the CPU. In FIG. 3, for convenience of illustration, one signal output line from the measurement sound processing unit 332 is shown, but in reality, there are signal output lines of measurement sound corresponding to the number of channels.

  The measurement sound signal output from the measurement sound processing unit 332 in the measurement processing block 33 is input to the power amplifier unit 4 via the switch 35 (Tm2 → Tm1), where it is amplified and output from the speakers 51 to 56. The When the measurement sound processing unit 332 outputs the sound signal of the measurement sound (phoneme) to a plurality of channels at the same time, the power amplifier unit 4 amplifies each of the individual measurement sound signals for each of these channels, Output from the speaker of the corresponding channel.

  Predetermined measurement sound signals uttered from the speakers are collected by the microphones 6a and 6b and input to the microphone amplifiers 34a and 34b. The microphones 6a and 6b are installed so that sound can be collected at a listening position (correction position) where the best correction sound field is desired in the listening environment. For example, as shown in FIG. 2, it may be installed at substantially the center of the listening environment. When the audio set 1 is an in-vehicle device, an optimal sound field is obtained when the user is listening at the driver's seat. The microphones 6a and 6b may be installed at the position of the user's ear seated in the driver's seat so as to be obtained.

  Ambient environmental sounds including measurement sounds are collected by the microphones 6a and 6b, amplified by the microphone amplifiers 34a and 34b, and input to the measurement units 331a and 331b of the measurement processing block 33. The measurement units 331a and 331b perform A / D conversion on the input audio signal, and perform various signal processing such as impulse response and frequency analysis processing using FFT on the obtained signal. As these processing results, in addition to information such as distances from the speakers of the respective channels to the installation positions of the microphones 6a and 6b, measurement results for items necessary for generating a sound field are obtained. The microphone characteristic correction unit 333 corrects the sound collection characteristic difference between the microphones 6a and 6b from the measurement results measured by the measurement units 331a and 331b.

  In order to correct the difference in sound collection characteristics, audio signals collected simultaneously by the microphones 6a and 6b that are equidistant from the speaker to be measured are required. Therefore, the control unit 7 determines whether or not the distance between each microphone measured by the measurement units 331a and 331b and the target speaker is within an error range, and a message that moves so as to be approximately the same distance as necessary. To the user. Further, as will be described later, when the support shaft of the microphone set including the microphones 6a and 6b is configured to be rotatable, a process of performing minute rotation driving so as to be approximately equidistant is executed.

  Here, as a specific example of the measurement processing by the measurement processing block 33, a configuration and operation for measuring the distance between each speaker on which the audio set 1 is arranged and the listening position, that is, the microphones 6a and 6b will be described.

  The distance between the speaker arranged in the listening environment and the listening position in the audio set 1 can be represented by information based on the arrival time to the listening position for each speaker corresponding to the audio channel. That is, the distance information from the speaker to the listening position can be converted into a time difference caused by the distance difference, and this delay time information can be used as a coefficient in the delay processing unit 321 of the sound field correction processing block 32. Correcting the arrival time difference caused by the difference in distance from the speaker to the listening position with the amount of time delay given at the time of occurrence from the speaker is sometimes referred to as time alignment.

  To generate a more realistic sound field at a listening point in the listening environment, it is necessary to adjust the time alignment at that point, but the microphone used to measure the information necessary to evaluate the sound field If there is a difference in characteristics, it is impossible to measure accurately. The audio set 1 can compensate for this difference in sound collection characteristics.

  As a method for measuring the distance from each speaker to the listening position, the following method can be given. First, measurement is performed sequentially for each speaker among a plurality of speakers provided in the audio set 1. A measurement sound signal is output from the speaker 51. As the measurement sound signal, a TSP (Time Stretched Pulse) signal having a predetermined frequency band characteristic can be used. The TSP signal is generated by the measurement sound processing unit 332 and collected by the microphones 6a and 6b installed corresponding to the listening position (that is, the correction position). The signal is input to the measurement unit 331 via the microphone amplifiers 34a and 34b. The measurement units 331a and 331b obtain sampling data extracted in units of a predetermined number of samples from the waveform of the input audio signal. A so-called impulse response is obtained by dividing the sampling data by the TSP signal on the frequency axis. The measurement units 331a and 331b can obtain distance information from the speaker to the listening position by executing predetermined signal processing or calculation processing for measurement based on the impulse response.

  The microphone characteristic correction unit 333 compares the characteristic information obtained from the impulse response obtained from the voice signal input from the microphone 6a with the characteristic information obtained from the impulse response obtained from the voice signal input from the microphone 6b, and collects sound for each microphone. Processing to compensate for the characteristic difference is performed.

  Thereafter, the microphone characteristic correction unit 333 performs a predetermined sound collection characteristic difference correction on the response signals measured by the microphones 6a and 6b and performs acoustic distance measurement using an impulse response. When the above processing is completed, more accurate distance information between the speakers and the microphones 6a and 6b can be obtained for each speaker that finally configures the plurality of audio channels of the audio set 1.

  Hereinafter, the sound field generation using the impulse response will be described. FIG. 4 shows a processing configuration for measuring the distance between the speaker and the microphone (listening position) by inputting the impulse response of the measurement sound signal generated by the measurement sound processing unit 332 in the measurement unit 331 of the measurement processing block 33. Is shown as a functional block. A process flow according to the configuration shown in FIG. 4 will be described with reference to FIGS.

  FIG. 5A shows an original waveform of an impulse response which is sampling waveform data. The horizontal axis of Fig.5 (a) shows the number of samples, and a vertical axis | shaft shows an amplitude level. FIG. 7 shows the frequency characteristics of the impulse response original waveform. The original waveform of the impulse response shown in FIG. 5A was obtained by performing a sampling process with 4096 samples. The number of samples 4096 is represented by 2 to the 12th power, and is set on the basis that the number of samples suitable for frequency analysis processing by FFT (Fast Fourier Transform) or the like is a power of 2. The sampling frequency fs = 48 kHz.

  In addition, as the sampling timing of the impulse response, the sample start time, that is, the timing when the sample point is 0 is set to coincide with the time when the measurement sound signal is started to be output from the measurement sound processing unit 332. In other words, the impulse response collected by the microphones 6a and 6b or the sampling timing of all the collected voice signals is made coincident with the time when the voice output from the speaker is started. Note that the measurement sound signal used for measuring the impulse response may be referred to as an impulse signal.

  FIG. 5B shows a waveform obtained by enlarging the rising position of the impulse response original waveform shown in FIG. 5A in the sample point direction (horizontal axis direction). The sample data of the impulse response original waveform shown in FIGS. 5A and 5B are input to the square processing unit 201 shown in FIG. 4 and branched to the frequency analysis / filter characteristic determining unit 202. Also input.

  The square processing unit 201 performs a square process on the amplitude value of the impulse response. As a result, the waveform data of the impulse response that originally has positive / negative amplitude values are squared as shown in FIG. 6A, and the negative amplitude value is inverted and folded to become the positive amplitude value. . As a result, the amplitude value of the negative electrode can be treated as an amplitude having the same polarity as that of the positive electrode in the subsequent processing, and only measurement for the positive electrode level is performed when measuring the impulse response amplitude value described later. I will do better. FIG. 6B shows a waveform obtained by enlarging the rising position of the impulse response original waveform shown in FIG. 6A in the sample point direction (horizontal axis direction).

  This sample data is sent to the variable low-pass filter 203. The variable low-pass filter 203 receives sample data of an impulse response based on a square sequence that is an output of the square processing unit 201. The variable low-pass filter 203 is provided in order to obtain an envelope waveform (envelope) suitable for a measurement target by removing high-frequency components to be treated as noise in the square response processed impulse response sample data (square waveform). However, depending on the filter characteristics, the entire envelope waveform including the rise of the impulse response may become too smooth. For this reason, the filter provided here is a variable low-pass filter that can be adaptively changed to the frequency characteristics of the impulse response.

  The frequency analysis / filter characteristic determination unit 202 performs frequency analysis on the input sampling data of the original waveform of the impulse response, for example, by FFT. Then, based on the frequency characteristic (frequency response) obtained by frequency analysis, the balance of the amplitude value of the middle frequency band and the high frequency band is determined, and the filter characteristic of the variable low-pass filter 203 is determined according to the determination result. The optimum value is determined as appropriate.

  The signal waveform that has passed through the variable low-pass filter 203 is shown in FIG. The sample data of the envelope shown in FIG. 8 is input to each of the delay sample number determination unit 204 and the threshold setting processing unit 205. The threshold value setting processing unit 205 obtains the peak level Pk from the sample data of the low-pass filter processing waveform shown in FIG. 8, and sets the amplitude level value obtained by a predetermined ratio with respect to the peak level Pk as the threshold value th. The threshold value setting processing unit 205 notifies the delay sample number determination unit 204 of the set threshold value th.

  The delay sample number determination unit 204 compares the amplitude value of the sample data of the signal waveform processed by the low-pass filter shown in FIG. 8 with the notified threshold th, and first performs a low-pass filter process starting from the sample point 0. A sample point whose waveform is equal to or greater than a threshold th is detected. In FIG. 8, this detected sample point is shown as a delayed sample point PD. The delay sample point PD represents the time delay from the sample point 0 corresponding to the start time of the sound output of the impulse signal from the speaker to the time point when the impulse response is assumed to be expressed by the number of samples. The variable low-pass filter 203 having an appropriate filter characteristic set by the control of the frequency analysis / filter characteristic determining unit 202 detects the delay sample point PD with high accuracy without causing an error.

  Information on the delay sample point PD determined by the sample number determination unit 204 as described above is notified to the spatial delay sample number calculation unit 206. The delay sample point PD samples the time delay until the point when the impulse response obtained by collecting the sound of the impulse signal with the microphone starts from the starting point of the sound output of the impulse signal from the speaker. It is expressed by numbers. That is, it can be considered that the distance between the speaker and the microphone is expressed in terms of time.

  However, in reality, a signal output system for outputting an impulse signal from a speaker, and a signal input until sampling is performed to collect sound output from the speaker by a microphone and obtain sample data of an impulse response original waveform In the system, there is a so-called system delay such as a filter delay, a processing delay caused by A / D or D / A conversion processing, or the like. The delay sample point PD determined by the sample number determination unit 204 includes an error due to a system delay or the like. Therefore, the spatial delay sample number calculation unit 206 subtracts an error caused by a system delay or the like from the delay sample point PD to thereby obtain a true delay sample number (hereinafter referred to as a space) corresponding to the speaker-microphone (listening position) distance. It is called the number of delay samples.) Information on the number of spatial delay samples obtained by the spatial delay sample number calculation unit 206 is notified to the distance calculation unit 207.

  The distance calculation unit 207 converts the notified number of spatial delay samples into time. Then, the distance between the speaker and the microphone is calculated using a predetermined arithmetic expression using the time-converted information on the number of spatially delayed samples and the value indicating the sound velocity. The calculated speaker-microphone distance information is stored in a nonvolatile memory or the like provided in the control unit 7 after the speaker to be measured is associated with the audio channel of the channel output from the speaker. Retained.

  Based on the difference between the speaker-microphone distances of each audio channel, the control unit 7 determines a spatial audio arrival time difference from the speaker of each audio channel to the listening position according to the distance difference. Based on the determination result, the delay processing unit 321 performs control for setting a predetermined delay constant for each audio channel so that the time difference between the sounds reaching the listening position from each speaker corresponding to the audio channel is eliminated. Is going. The delay processing unit 321 executes delay processing for each audio signal set by the control unit 7. As a result, at the appropriate listening position, a sound field is generated in which the variation in the arrival time of the sound due to the difference in the distance between the speaker and the listening position is canceled. That is, a sound field in which time alignment is appropriately corrected at the listening position is generated.

  In the sound field generation processing described above, a specific method for compensating for the difference in sound collection characteristics of the microphones 6a and 6b will be described.

  The listening environment of this specific example is a case where the microphones 6a and 6b are installed at positions that are not so far away from the speaker in an automobile or a small room. It can be assumed that the difference in the sound collection characteristics due to the state in the listening environment such as standing wave and reflection by the wall is very small. Further, the microphone 6a and the microphone 6b are fixed to each other with an interval such that a characteristic difference depending on the installation position does not appear in the listening environment.

  In this specific example, the impulse response, frequency amplitude response, time alignment, and the like at the microphone installation position described with reference to FIGS. 5 to 7 are calculated in two ways corresponding to the microphones 6a and 6b.

  Since the audio set 1 of this specific example corresponds to the 5.1 channel surround system and has a center speaker, in the sound field measurement mode, the user uses a microphone system in which the microphone 6a and the microphone 6b are connected. What is necessary is just to install in the arrangement | positioning as shown in FIG. 2 toward a center speaker. A suitable viewing position of the user to be corrected is approximately in the center of the area surrounded by the speakers. In the audio set 1, the center speaker is arranged in the approximate center of the region, that is, substantially in front of the user's preferred viewing position. Therefore, when the microphone set is arranged as shown in FIG. The distance is approximately equidistant.

  When the impulse response of the speaker 51 (here, the center speaker) with respect to the microphones 6a and 6b is measured with this microphone arrangement, and further the frequency amplitude analysis such as FFT is performed on the impulse response waveform, the frequency amplitude response is calculated. The microphone characteristic correction unit 333 compensates for the sound collection characteristic difference between the microphone 6a and the microphone 6b.

  As described with reference to FIGS. 5 to 8 described above, the audio set 1 uses the TSP signal as the measurement sound signal to calculate the impulse response between the speaker and the microphone. Based on the rise of this signal, the distance between the speaker and each microphone is calculated. The rising threshold value of the impulse response is determined from the maximum peak value of the impulse response.

  9A shows an impulse response of the measurement sound signal collected by the microphone 6a, and FIG. 9B shows an impulse response of the measurement sound signal collected by the microphone 6b.

  The microphone characteristic correction unit 333 determines the rising time from the impulse response original waveform, and determines whether the rising times measured by the two microphone devices are the same or within a predetermined allowable time. When they are the same or within the allowable time, it can be determined that the distance between the speaker and the microphone 6a and the distance between the speaker and the microphone 6b are substantially equal. If the microphone 6a and the microphone 6b are not equidistant from the center speaker, it may be presented to the user that the microphones 6a and 6b are equidistant from the center speaker.

  When the microphone 6a and the microphone 6b are equidistant from the speaker 51, the frequency response of the impulse response collected by the microphone 6a and the frequency response of the impulse response collected by the microphone 6b Then, an average value of gains in this window is calculated using a specific band as a window.

  10A shows the frequency amplitude response of the impulse response original waveform of the measurement sound signal collected by the microphone 6a, and FIG. 10B shows the impulse response source of the measurement sound signal collected by the microphone 6b. The frequency amplitude response of the waveform is shown.

  For example, the sampling width of the frequency amplitude response is f0, f1,..., F (n−1), and the frequency amplitude level at each sample point is A0, A1,. Is calculated by FFT. At this time, as shown in FIG. 10, the observation window is set to frequency bands fa to fb (however, f0 ≦ fa and fb ≦ f (n−1)). At this time, the dB average value of the gain is obtained by the following equation (1).

  Here, equation (1) is one method of gain average calculation, but the calculation method is not limited to equation (1). The specific band for which the gain average is calculated here is several hundred to several kilohertz, which is known to have relatively small variations in the frequency of sound signals collected by the microphones 6a and 6b to be used. Preferably, the frequency band is

  For example, as shown in FIG. 9, it can be considered that the rise times of the impulse responses of the microphones 6a and 6b are sufficiently the same within a certain range, but the gain of the impulse response waveform is lower in the microphone 6b than in the microphone 6a. Is low, the average gain value for each microphone in the frequency band fa to fb is obtained for the audio signals input from the microphones 6a and 6b. Then, in the subsequent time alignment correction processing for each speaker constituting the audio channel, the level of the microphone having the low gain average value is matched with the level of the microphone having the high gain average value based on the obtained average gain value. The level correction is performed to compensate for the difference in sound collection characteristics between microphones.

  The characteristic difference between the microphones 6a and 6b may be corrected other than the gain difference (sensitivity difference) described above. For example, the frequency amplitude characteristic may be corrected by adding the characteristic difference between the frequency amplitude characteristics (dB value) between the microphone 6a and the microphone 6b to the microphone with the lower gain. Of course, the gain corresponding to the characteristic difference may be subtracted from the microphone having the higher gain. In this case, after determining that the distance between both microphones and the speaker is the same, the microphone characteristic correction unit 333 compares the characteristic difference of the frequency amplitude response calculated from the impulse response, and is indicated by a dotted line in FIG. The characteristic difference of the frequency amplitude characteristic (dB value) between the microphone 6a and the microphone 6b is made to the microphone with the lower gain so that it approximates the sound collection characteristic waveform in the microphone 6a shown in FIG. to add. Alternatively, the gain corresponding to the characteristic difference is subtracted from the microphone with the higher gain. This correction is advantageous in calculations such as a frequency band that can be faithfully reproduced, so-called “f-special” correction.

  Further, when the microphone characteristic correcting unit 333 determines the rise time from the impulse response original waveform, as a countermeasure when a speaker that is equidistant from the microphone 6a and the microphone 6b cannot be found, the characteristic depending on the installation position of both in the listening environment For example, the microphone 6a and the microphone 6b, which are fixed to each other with an interval that does not cause a difference, can be rotated with respect to the speaker.

  FIG. 12 shows a microphone set in which speakers 61 to 65 in which each channel is arranged at a rough position in a listening environment and microphones 6a and 6b arranged at arbitrary positions are fixed to each other and turnable. The positional relationship with 60 is shown. In this case, the control unit 7 sends a signal for rotating the microphone set 60 according to the result of the rise time determination of the impulse response original waveform.

  First, it is assumed that the angle of the microphone set 60 with respect to the speaker 63 is in the state St1. At this time, the microphone characteristic correction unit 333 determines the rise time from the impulse response original waveform when the measurement sound signal uttered from the speaker 63 is collected by the microphone 6a and the microphone 6b. Upon receipt of this determination result, the control unit 7 sends a signal for turning the microphone set 60 by a small angle in any direction when a speaker that is equidistant from the microphones 6a and 6b cannot be found. Transition to St2. This is repeated to search for a speaker in which both the microphone 6a and the microphone 6b are equidistant.

  In the case of such a mechanism, the sound collection characteristics and the sound field measurement are executed after adjusting the direction of the microphones 6a and 6b with respect to the speaker to the optimum positions that are equidistant from each other by changing the rotation angle of the microphone system itself. Therefore, it is possible to generate an audio signal based on a result measured under a condition in which the distances between the microphones and the speakers are exactly the same. In addition, the sound field created by uttering the sound signal generated in this way from each speaker increases the sense of reality at the appropriate listening position and improves the reality.

  As described above, in the audio set 1, the microphone characteristic correction unit 333 can compare the impulse responses for the individual microphones measured by the measurement units 321a and 321b and correct the sound collection characteristic difference between the two microphones. When the optimum signal processing is performed on the audio signals of the respective channels based on the parameters analyzed from the sound signals collected by the microphones 6a and 6b, more strict correction can be performed. The sound field created in the listening environment by the sound signal corrected in this way is more realistic for the listener at the appropriate listening position, and the reality is improved.

  The audio set to which the sound field measuring apparatus 1 is applied may be an AV (Audio video) system capable of reproducing not only audio but also video. In this case, the audio set includes a liquid crystal display (LCD) as display means for displaying video data, and a configuration capable of reproducing video content data. As a reproduction environment, the present invention can be applied to a car interior, a room other than a car, and other audio listening environments.

  Furthermore, in the above description, the correction information is the propagation delay time from the speaker to the listening position, and the example in which the sound field correction is adjustment of time alignment (signal delay time adjustment) has been described. The sound field correction with respect to the target correction position based on this may be, for example, equalizing correction in the equalizer unit 322 in FIG. 2 or volume correction in the gain adjusting unit 323 in addition to time alignment. Also, a plurality of these correction elements can be used in combination.

It is a block diagram explaining the audio set to which the sound field measuring device according to the present invention is applied. It is the schematic explaining the arrangement | positioning of the speaker and microphone in the said audio set. It is a block diagram explaining the sound field correction / measurement function unit in the audio set. It is a functional block diagram explaining the process for inputting the impulse response of the measurement sound signal and measuring the distance between the speaker and the microphone (listening position) in the measurement processing block of the sound field correction / measurement function unit. (A) is a waveform diagram showing the original waveform of the impulse response, (b) is a waveform diagram showing the rising position of the impulse response original waveform shown in (a) in the horizontal axis direction. (A) is a waveform diagram obtained by squaring impulse response waveform data having both positive and negative amplitude values, and (b) shows the rise position of the impulse response original waveform shown in (a) in the horizontal axis direction. It is a wave form diagram expanding and showing. It is a frequency characteristic figure which shows the frequency characteristic of an impulse response original waveform. It is a wave form diagram which shows the signal waveform which passed the variable low-pass filter in the measurement process block of the said sound field correction / measurement function part. (A) is a waveform diagram showing an impulse response of a measurement sound signal collected by one microphone of the audio set, and (b) is a measurement sound signal collected by the other microphone of the audio set. It is a wave form diagram which shows the impulse response. (A) is a waveform diagram showing a frequency amplitude response of an impulse response of a measurement sound signal collected by one microphone of the audio set, and (b) is a sound collected by the other microphone of the audio set. It is a wave form diagram which shows the frequency amplitude response of the impulse response of the measured sound signal. (A) is a waveform diagram showing a frequency amplitude response of an impulse response of a measurement sound signal collected by one microphone of the audio set, and (b) is a sound collected by the other microphone of the audio set. It is a wave form diagram which shows the waveform which added the characteristic difference to the frequency amplitude response of the impulse response of the measured sound signal. It is the schematic explaining the positional relationship of the speaker and microphone set which are arrange | positioned at listening environment.

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 Audio set, 2 Media reproduction part, 3 Sound field correction unit, 4 Power amplifier part, 5 Speaker, 6a, 6b Microphone, 7 Control part, 8 Memory part, 9 User interface part, 31 Sound field correction / measurement function part, 32 sound field correction processing block, 33 measurement processing block, 34a, 34b microphone amplifier, 35 switch, 331a, 331b measurement unit, 332 measurement sound processing unit, 333 microphone characteristic correction unit, 321 delay processing unit, 322 equalizer unit, 323 gain Adjustment section

Claims (7)

In a sound field measuring apparatus for measuring a sound field generated in a reproduction environment by an audio signal output from these speakers in an audio set including a plurality of speakers,
A first microphone that collects audio signals that are installed at arbitrary positions in the reproduction environment and output from the plurality of speakers;
A second microphone installed near the first microphone for collecting audio signals output from the plurality of speakers;
A measurement unit for measuring sound collection characteristics of the audio signal in the first microphone and sound collection characteristics of the audio signal in the second microphone;
A characteristic analyzer for analyzing the measured sound collection characteristics;
The difference in sound collection characteristics between the first microphone and the second microphone based on the analysis value analyzed by the characteristic analysis unit with respect to the sound signals collected by the first microphone and the second microphone. And a correction value generation unit that generates a correction value for performing correction to reduce the sound field.   The characteristic analysis unit analyzes the distance between the first microphone and the speaker that is the measurement target among the plurality of speakers and the distance between the second microphone and the speaker that is the measurement target, and the distance between the two is When the distance is approximately equal, correction is performed to reduce the difference in sound collection characteristics between the first microphone and the second microphone with respect to the sound signals collected by the first microphone and the second microphone. The sound field measuring device according to claim 1.   3. The sound field measuring apparatus according to claim 2, wherein the first microphone and the second microphone are fixed to each other at an interval that does not cause a characteristic difference depending on an installation position in the reproduction environment.   2. The sound field measuring apparatus according to claim 1, wherein the sound collection characteristic is a characteristic of sound collection sensitivity in the first microphone or the second microphone.   2. The sound field measuring apparatus according to claim 1, wherein the sound collection characteristic is a frequency amplitude characteristic of an audio signal collected by the first microphone or the second microphone.   The sound field measurement device according to claim 1, wherein the sound collection characteristic is calculated by an impulse response in the first microphone or the second microphone. In a sound field measurement method for measuring a sound field generated in a reproduction environment by an audio signal output from these speakers in an audio set including a plurality of speakers,
A first sound collecting step of collecting an audio signal output from a predetermined speaker among the plurality of sounds by a first microphone installed at an arbitrary position in the reproduction environment;
A second sound collecting step of collecting an audio signal output from the predetermined speaker by a second microphone installed in the vicinity of the first microphone;
A measurement step of measuring the sound collection characteristics of the sound signal acquired in the first sound collection step and the sound collection characteristics of the sound signal acquired in the second sound collection step;
A characteristic analysis step for analyzing the measured sound collection characteristic;
Based on the analysis value analyzed in the characteristic analysis step on the audio signal collected in the first sound collection step and the second sound collection step, the first microphone and the second microphone A correction value generating step for generating a correction value for performing correction to reduce the sound collection characteristic difference,
In the characteristic analysis step, the distance between the first microphone and the speaker that is the measurement target among the plurality of speakers and the distance between the second microphone and the speaker that is the measurement target are analyzed, and the distance between the two is approximately When the distance is equal, correction is performed to reduce a difference in sound collection characteristics between the first microphone and the second microphone with respect to the sound signals collected by the first microphone and the second microphone. Characteristic sound field measurement method.

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