JP4408909B2 - Sound field correction method in audio system - Google Patents

Description

  The present invention relates to a sound field correction method for correcting sound field characteristics in an audio system.

  An audio system is required to create a sound field space that provides a sense of realism, and a sound field correction method for an audio system disclosed in Japanese Utility Model Laid-Open No. 6-13292 is conventionally known.

  This conventional audio system includes an equalizer for adjusting the frequency characteristics of an input audio signal, and a delay circuit that delays an audio signal output from the equalizer, and an output of the delay circuit is supplied to a speaker. It has a configuration.

  Further, in order to correct the sound field characteristics, a pink noise generator, an impulse generator, a selector circuit, a microphone for measuring a reproduced sound reproduced by a speaker, a frequency analysis unit, and a delay time calculation unit are provided. Is provided. The pink noise generated by the pink noise generator is supplied to the equalizer via the selector circuit, and the impulse signal generated by the impulse generator is directly supplied to the speaker via the selector circuit.

  When correcting the phase characteristics of the sound field space, the impulse signal is directly supplied from the impulse generator to the speaker, the impulse sound reproduced by the speaker is measured by the microphone, and the delay time calculating means calculates the measurement signal. By analyzing, the propagation delay time of the impulse sound from the speaker to the listening position is measured.

  That is, the impulse signal is directly supplied to the speaker, and the time difference between the time when the impulse signal is supplied to the speaker and the time when the impulse sound reproduced by the speaker reaches the microphone is obtained by the delay time calculating means, The propagation delay time is measured. Then, the phase characteristic of the sound field space is corrected by adjusting the delay time of the delay circuit based on the measured propagation delay time.

  Also, when correcting the frequency characteristics of the sound field space, pink noise is supplied from the pink noise generator to the equalizer, and the pink noise reproduction sound reproduced by the speaker is measured with a microphone, and the frequency characteristics of the measurement signal are determined. The frequency analysis means is used for analysis. The frequency characteristic of the sound field space is corrected by feedback control of the frequency characteristic of the equalizer based on the analysis result.

Japanese Utility Model Publication No. 6-13292

  However, in the conventional audio system, as described above, when correcting the phase characteristic of the sound field space, the impulse signal is directly supplied to the speaker. Therefore, there is a problem that the phase characteristic of the entire audio system is not corrected to the phase characteristic that generates an appropriate sound field space.

  Further, when correcting the frequency characteristics of the sound field space, a method is adopted in which the frequency characteristics of the pink noise reproduction sound are analyzed using a narrow band filter group, and the analysis result is fed back to the equalizer.

  However, when the frequency characteristics of the measurement signal obtained from the pink noise reproduction sound reproduced by the speaker in this way is analyzed with the individual narrowband filters of the narrowband filter group, an accurate analysis that matches the frequency characteristics of the equalizer Since a result cannot be obtained, there is a problem that it is difficult to appropriately correct the frequency characteristic of the sound field space if the frequency characteristic of the equalizer is food-back controlled based on the analysis result.

  It is an object of the present invention to provide a sound field correction method for overcoming the above-described problems of the prior art and realizing a higher quality sound field space.

According to the first aspect of the present invention, each of a plurality of channels includes a signal processing system that performs signal processing on an audio signal and sound emission means that reproduces sound based on an output of the signal processing system. The signal processing system includes a frequency dividing unit that is a plurality of filters that discriminate and output the audio signal into a plurality of frequency bands, and an attenuator that performs a plurality of gain adjustments provided in association with the plurality of filters. An interband level adjusting means, an adding means for generating an added signal by adding the signals discriminated by the frequency dividing means and gain-adjusted by the interband level adjusting means, and gain adjustment for the added signal An inter-channel level adjusting means for applying a delay, and a delay for performing a delay process on the signal whose gain has been adjusted by the inter-channel level adjusting means and outputting to the sound emitting means And a stage are provided, a sound field correcting method in an audio system, in the signal processing system of all channels equal the gain of a plurality of attenuators is the inter-band level adjusting means and the channel-to-channel level adjusting equal delay time equal to and the delay means the gain of the means, further to supply noise exclusively and sequentially to the signal processing system for each channel, and the noise is provided in the supplied signal processing system The noise is divided into the frequency bands for each of the filters that are conducted in the order and exclusively by making each of the plurality of filters, which are frequency dividing means, in order and exclusively conducted. said listening noise tone played by the speaker provided in the channel of the signal processing system for outputting the noise By detecting at location, a first detection step of detecting a total number amount of noise sound filter provided in the signal processing system of all the channels, each noise sound detected by said first detection step The sound pressure is compared with a predetermined threshold, and a sound emitting means that reproduces a noise sound having a sound pressure larger than the threshold is determined and a sound emitting means that reproduces a noise sound having a sound pressure smaller than the threshold, A plurality of attenuators , which are channel level adjustment means for channels provided with sound emission means for reproducing the determined noise sound having a large sound pressure, are gain- adjusted according to a predetermined target curve, and the noise sound having the determined low sound pressure is determined. The signal processing system for each channel is obtained by not adjusting the gain of a plurality of attenuators which are level adjusting means between the bands of the channels provided with the sound emitting means for reproducing the sound. A frequency characteristic correcting step for setting frequency characteristics by the frequency dividing means and the interband level adjusting means, and a signal processing system for each channel for which the frequency characteristics are set in the frequency characteristic correcting step is exclusive for each channel. In addition, by selecting in order and simultaneously supplying noise to a plurality of filters that are frequency division means provided in the signal processing system of the selected channel, the frequency characteristics of the noise are compared with the plurality of filters. The noise sound reproduced by the sound emitting means provided in the channel of the selected signal processing system is set by a plurality of attenuators which are level adjustment means between the bands provided in the signal processing system of the selected channel . a second detection step of detecting the sound pressure at the listening position, each channel in the second detection step And between channels to adjust the gain of the data level correcting step - multiple attenuator is issued the channel-to-channel level adjusting means based on a minimum value of sound is provided in the signal processing system of each channel of the noise sound The signal processing system of each channel for which the frequency characteristics are set and the gain is adjusted in the frequency characteristic correction step and the interchannel level correction step is selected exclusively and sequentially for each channel, and the selected channel is selected. noise signal processing supplying the noise simultaneously for the plurality of filters is a frequency division means provided in the system, is reproduced on the basis of the noise output from the signal processing system of channel that the selected to sound emission means a third detection step of detecting a sound by the listening position, each detected for each of the channels in the third detection step Neu Phase characteristics for adjusting the delay time of the delay means provided in the signal processing system of each channel to a delay time corresponding to the distance from the sound emission means of each channel to the listening position based on the phase difference of the sound And a correction step.

The invention according to claim 2 is a signal processing system (hereinafter referred to as a “low-frequency channel signal processing system”) that performs signal processing on a low frequency range of an audio signal and a sound based on the output of the low frequency channel signal processing system. And a sound emission means for reproducing the signal, a signal processing system for performing signal processing on the entire area of the audio signal (hereinafter referred to as “global channel signal processing system”), and the global channel signal processing system Sound output means for reproducing sound based on the output is provided in each of the plurality of all-range channels, and the signal processing system of each of the all-channel channels discriminates and outputs the entire range of the audio signal into a plurality of frequency bands. is attenuator performing a first frequency dividing means is a plurality of filters, a plurality of gain adjustment provided in association with the plurality of filters 1 and inter-band level adjustment means, first to generate a first sum signal by adding the signal gain has been adjusted by the first are discriminated by frequency division means and the first inter-band level adjusting means Adding means, first inter-channel level adjusting means for adjusting the gain for the first added signal, and delay processing for the signal subjected to gain adjustment by the first inter-channel level adjusting means. And a first delay means for outputting to the sound emitting means, and the low-frequency channel signal processing system includes a plurality of filters that discriminate and output the low frequency of the audio signal into a plurality of frequency bands. A second frequency dividing means , a second interband level adjusting means that is an attenuator for performing a plurality of gain adjustments provided in association with a plurality of filters that are the second frequency dividing means , and the second Frequency A second adding means for generating a second sum signal are discriminated by the dividing means by adding the signal gain has been adjusted by the second inter-band level adjusting means, the gain with respect to the second addition signal A second inter-channel level adjusting unit that performs adjustment; and a second delay unit that performs delay processing on the signal that has been gain-adjusted by the second inter-channel level adjusting unit and outputs the result to the sound emitting unit. A method for correcting a sound field in an audio system, comprising: a plurality of attenuators serving as the first and second inter-band level adjusting means in all the all-band channel signal processing system and low-band channel signal processing system said equal gain and first, equally and the first gain of the second channel-to-channel level adjusting means, and equalizing the delay time of the second delay means, further wherein the noise throughout Ji Each signal processing system of the tunnel and the low channel is supplied to the exclusive and order, and exclusively the plurality of each filter is a frequency dividing means to which the noise is provided in the supplied signal processing system in order The noise emission is provided in the channel of the signal processing system that discriminates the noise into the frequency band for each filter that is conducted in the order and exclusively, and outputs the discriminated noise . A first detection step of detecting noise sounds corresponding to the total number of filters provided in the signal processing systems of all channels by detecting noise sounds reproduced by the means at the listening position; The sound pressure of each noise sound detected in the detection step is compared with a predetermined threshold value, and the sound emission means that reproduces the noise sound having a sound pressure value greater than the threshold value is smaller than the threshold value. A sound emitting means for reproducing the noise sound having the determined large sound pressure among the first and second inter-band level adjusting means is provided. The gain of the channel inter-band level adjusting unit is adjusted according to a predetermined target curve, and the gain of the channel inter-band level adjusting unit provided with the sound emitting unit that reproduces the noise sound having the determined small sound pressure is adjusted. by not adjusting the frequency characteristic correcting step of setting a frequency characteristic by said frequency division means and the inter-band level adjustment means in the signal processing system of each channel, the frequency characteristic in the frequency characteristic correcting process has been set the each entire channel signal processing system selects the exclusive and sequentially for each channel, provided in the selected whole channel signal processing system By supplying the noise simultaneously to a plurality of filters is a first frequency division means have a first provided a frequency characteristic of the noise to the plurality of filters and the selected entire channel signal processing system A sound pressure of a noise sound that is set by a plurality of attenuators that are level adjustment means between bands and that is reproduced based on noise that is output from the selected global channel signal processing system to the sound emission means of the channel is the listening position. And a first detection step provided in each of the global channel signal processing systems based on the minimum value of the sound pressure of the noise sound detected for each global channel in the second detection step. a channel-to-channel level correcting step of adjusting the gain of a plurality of attenuators is channel-to-channel level adjusting means, the frequency characteristic correcting Engineering And a setting and wherein said gain adjustment is set in the frequency characteristic in the frequency characteristic correcting step and the entire channel signal processing system has been low-frequency channel signal processing system of the frequency characteristic in the channel-to-channel level correcting step Are selected for each channel exclusively and sequentially, and noise is simultaneously supplied to a plurality of filters as frequency dividing means provided in the signal processing system of the selected channel, and the signal of the selected channel is selected. A third detection step of detecting the sound pressure of the noise sound reproduced based on the noise output from the processing system to the sound output means of the channel at the listening position; and for each channel in the third detection step Based on the detected phase difference of each noise sound, the first delay means provided in each of the all-range channel signal processing systems and the low-frequency channel And phase characteristic correcting step of adjusting the delay time of the second delay means provided in the tunnel signal processing system, the delay time corresponding to the distance from sound emission unit of each channel to the listening position, the phase After the characteristic correction step, the noise is simultaneously supplied to all the all-channel signal processing systems, and the low-frequency filter is provided among the plurality of filters that are the first frequency division means to which the noise is supplied. The attenuator of the interband level adjusting means is non-conductive, the attenuator of the interband level adjusting means provided in the filter relating to the mid-high range is made conductive, and the sound emission provided in all the global channel signal processing systems detecting a medium and high frequency noise tone played simultaneously by means the listening position, then, to simultaneously supply noise to all of the all-channel signal processing system Together provided an attenuator inter-band level adjusting means for the noise is provided in the filter according to the low frequency of the plurality of filters is a respective first frequency dividing means supplied conduction, the filter according to the mid-high range The attenuator of the inter-band level adjusting means is turned off, and the noise sound simultaneously reproduced by the sound emitting means provided in all the all-channel signal processing systems is used as the first low-frequency noise sound. detected by the position, then, the simultaneous supply noise to a plurality of filters is a second frequency dividing means in the low-frequency channel signal processing system, with sound emitting means provided on the low-frequency channel signal processing system A fourth detection step of detecting the reproduced noise sound as a second low-frequency noise sound at the listening position; and the medium detected in the fourth detection step On the basis of the ratio of the low frequency band spectrum of the mid-high band spectrum and the first low-frequency noise sound low-frequency spectrum and the second low-frequency noise sound relative difference of the frequency noise sound, the low-frequency channel signal processing system By adjusting the gains of a plurality of attenuators which are second channel level adjusting means provided in the sound field space reproduced by the sound emission means of the entire channel and the sound emission means of the low frequency channel And a flattening correction step of flattening the frequency characteristics of the reproduced sound.

According to a third aspect of the present invention, in the sound field correction method according to the first or second aspect, a noise sound generated from the sound emitting means is detected a plurality of times by repeating the first step a plurality of times, and the frequency In the characteristic correction step, an average value of sound pressure for each channel is calculated based on the sound pressure of each noise sound detected in the first detection step, and the average value of sound pressure for each channel and the sound pressure The determination is performed by comparing with a threshold value .

According to a fourth aspect of the present invention, in the sound field correction method according to the first or second aspect, a noise sound generated from the sound emitting means is detected a plurality of times by repeating the second detection step a plurality of times,
In the inter-channel level correcting step, the gain of each inter-channel level adjusting means is adjusted based on the average value of the noise sound detected a plurality of times in the second detecting step .

According to a fifth aspect of the present invention, in the sound field correction method according to the first or second aspect, a noise sound generated from the sound emitting means is detected a plurality of times by repeating the third detection step a plurality of times, In the phase characteristic correction step, the delay time of each delay means is adjusted based on the phase difference obtained based on the average value of the noise sound detected a plurality of times in the third detection step. To do.

The invention according to claim 6 is the sound field correction method according to claim 1 or 2, wherein the audio system is a 5.1 channel multi-channel audio system .

  According to the sound field correction method of the present invention, the sound field correction is performed in consideration of the characteristics of the audio system and the speaker, and the sound field correction is performed under the same conditions as in audio reproduction. The sound field correction is performed in consideration of the characteristics of the sound field and the sound field environment. In addition, it is possible to provide a very high-quality and realistic sound field space.

  In addition, since the level of the playback sound is flat over the entire audio frequency band, it prevents the generation of playback sound that is annoying due to the level of a certain frequency in the audio frequency band becoming stronger or weaker, and is realistic. A sound field space can be provided.

  Embodiments of an automatic sound field correction system according to the present invention will be described below with reference to the drawings. FIG. 1 is a block diagram illustrating a configuration of an audio system including the automatic sound field correction system according to the present embodiment, and FIGS. 2 to 4 are block diagrams illustrating a configuration of the automatic sound field correction system.

  In FIG. 1, the audio system includes digital audio signals SFL, SFR, SC, SRL from a sound source 1 such as a CD (Compact disk) player or a DVD (Digital Video Disk or Digital Versatile Disk) player through a signal transmission path of a plurality of channels. , SRR, SWF, and a noise generator 3 are provided.

  Furthermore, D / A converters 4FL, 4FR, 4C, 4RL, 4RR, 4WF for converting the digital outputs DFL, DFR, DC, DRL, DRR, DWF, which have been signal-processed for each channel by the signal processing circuit 2, into analog signals; , Amplifiers 5FL, 5FR, 5C, 5RL, 5RR, and 5WF for amplifying the analog audio signals output from these D / A converters are provided. The analog audio signals SPFL, SPFR, SPC, SPRL, SPRR, SPWF amplified by these amplifiers are used as a plurality of channels of speakers 6FL, 6FR, 6C, 6RL, 6RR arranged in a listening room 7 as shown in FIG. , 6WF is supplied to sound.

  In addition, the microphone 8 that collects the reproduced sound at the listening position RV, the amplifier 9 that amplifies the sound collection signal SM output from the microphone 8, and the output of the amplifier 9 are converted into digital sound collection data DM for signal processing. An A / D converter 10 that supplies the circuit 2 is provided.

  Here, this audio system has frequency characteristics for reproducing only so-called deep bass sounds and full-band speakers 6FL, 6FR, 6C, 6RL, 6RR having frequency characteristics that can be reproduced over almost the entire audio frequency band. By sounding the low frequency reproduction dedicated speaker 6WF, a sound field space with a sense of reality is provided to the listener at the listening position RV.

  For example, as shown in FIG. 7, the listener arranges front left and right channel front speakers (front left speaker, front right speaker) 6FL, 6FR and a center speaker 6C in front of the listening position RV according to his / her preference. When the left and right two-channel surround speakers (rear left speaker, rear right speaker) 6RL and 6RR are arranged behind the listening position RV, and the subwoofer 6WF dedicated to low frequency reproduction is arranged at an arbitrary position, this audio system The automatic sound field correction system provided for the analog audio signals SPFL, SPFR, SPC, SPRL, SPRR, and SPWF whose frequency characteristics and phase characteristics are corrected to these six speakers 6FL, 6FR, 6C, 6RL, 6RR, and 6WF. The sound field space with a sense of reality is realized by supplying the sound to the sound.

  The signal processing circuit 2 is formed by a digital signal processor (DSP) or the like. The digital signal processor or the like constitutes an automatic sound field correction system that performs sound field correction in cooperation with the noise generator 3, the amplifier 9, and the A / D converter 10.

  That is, the signal processing circuit 2 includes system circuits CQT1, CQT2, CQT3, CQT4, CQT5, CQTk having substantially the same configuration provided in the signal transmission path of each channel shown in FIG. 2, and the frequency characteristic correction unit shown in FIG. 11, an inter-channel level correction unit 12, a phase characteristic correction unit 13, and a flattening correction unit 14 are provided. In the automatic sound field correction system, the frequency characteristic correction unit 11, the interchannel level correction unit 12, the phase characteristic correction unit 13, and the flattening correction unit 14 control the system circuits CQT1, CQT2, CQT3, CQT4, CQT5, and CQTk. It is configured. In the following description, each channel is indicated by a number x (1 ≦ x ≦ k).

  The configuration of the system circuit CQT1 provided in the first channel (x = 1) will be described as a representative. The switch element SW12 that controls on / off of the input of the digital audio signal SFL from the sound source 1, and the noise generator 3 is provided with a switch element SW11 for ON / OFF control of the input of the noise signal DN from 3. The switch element SW11 is connected to the noise generator 3 through the switch element SWN.

  Here, the switch elements SW11, SW12, SWN are controlled by a system controller MPU formed by a microprocessor described later. At the time of audio reproduction, the switch element SW12 is turned on (conductive), the switch elements SW11 and SWN are turned off (non-conductive), and at the time of sound field correction, the switch element SW12 is turned off and the switch elements SW11 and SWN are turned on.

  Band pass filters BPF11 to BPF1j are connected in parallel to the output contacts of the switch elements SW11 and SW12 as frequency discriminating means for a plurality of j, and the frequency division for dividing the frequency of the input signal by these bandpass filters BPF11 to BPF1j as a whole. Means are configured.

  Suffixes 11 to 1j attached to BPF11 to BPF1j indicate the order of the center frequencies f1 to fj of the bandpass filters BPF11 to BPF1j in the first channel (x = 1).

  Attenuators ATF11 to ATF1j called interband attenuators are connected to output contacts of the bandpass filters BPF11 to BPF1j, respectively. Accordingly, the interband attenuators ATF11 to ATF1j serve as transmission line level adjusting means for adjusting the levels of the outputs of the bandpass filters BPF11 to BPF1j.

  Further, each bandpass filter BPF11 to BPF1j is provided with an interband attenuator ATF11 to ATF1j, so that a variable gain type frequency discriminating means is configured by the bandpass filter and the interband attenuator corresponding to each other. That is, BPF11 and ATF11 are the first variable gain type frequency discriminating means, BPF12 and ATF12 are the second variable gain type frequency discriminating means, and BPF1j and ATF1j are the jth variable gain type variable gain type frequency discriminator. It is a means.

  An adder ADD1 is connected to the output contacts of the interband attenuators ATF11 to ATF1j, an attenuator ATG1 called an interchannel attenuator is connected to the output contact of the adder ADD1, and a delay is connected to the output contact of the interchannel attenuator ATG1. A circuit DLY1 is connected. The output DFL of the delay circuit DLY1 is supplied to the D / A converter 4FL in FIG.

  Here, each of the bandpass filters BPF11 to BPF1j is formed of a narrow band-pass type second-order Butterworth filter set to the center frequencies f1 and f2 to fi to fj, respectively, as shown in the frequency characteristic diagram of FIG. ing.

  That is, the bandpass filter which uses the respective frequencies f1, f2 to fi to fj as the respective center frequencies by dividing the entire frequency band of the speaker 6FL which can be reproduced from the low range to the mid to high range by an arbitrary number j. BPF11 to BPF1j are provided. Specifically, the low frequency band of about 0.2 KHz or less is divided into about 6, and the middle and high frequency band of about 0.2 KHz or more is divided into about 7, and the center frequency of each divided narrow frequency range is determined. The center frequencies f1, f2 to fi to fj of the bandpass filters BPF11 to BPF1j are used. Furthermore, by setting so that there is no gap between the pass frequency bands of the band pass filters BPF11 to BPF1j and the pass frequency bands do not substantially overlap, the entire frequency band is covered without omission. Yes.

  Further, the band pass filters BPF11 to BPF1j can be switched between conduction / non-passage exclusively under the control of the system controller MPU. Further, at the time of audio reproduction, the bandpass filters BPF11 to BPF1j are switched so as to be all conductive.

  The attenuators ATF11 to ATF1j are formed of digital attenuators and change the attenuation rate in the range from 0 dB to the minus side in accordance with the adjustment signals SF11 to SF1j from the frequency characteristic correction unit 11.

  The adder ADD1 adds the signals passing through the bandpass filters BPF11 to BPF1j and attenuated by the attenuators ATF11 to ATF1j, and supplies the added signal to the attenuator ATG1.

  The inter-channel attenuator ATG1 is formed of a digital attenuator, and the details thereof will be described in the description of the operation. However, the attenuation rate is changed in the range from 0 dB to the minus side in accordance with the adjustment signal SG1 from the inter-channel level correction unit 12.

The delay circuit DLY1 is formed of a digital delay circuit, and changes the delay time according to the adjustment signal SDL1 from the phase characteristic correction unit 13.
The system circuits CQT2, CQT3, CQT4, and CQT5 of the remaining channels x = 2 to 5 have the same configuration as the system circuit CQT1.

  That is, although simplified in FIG. 2, the center frequency f1 to fj is set in the system circuit CQT2 of the second channel (x = 2) following the switch elements SW21 and SW22. And j band-pass filters BPF21 to BPF2j, and interband attenuators ATF21 to ATF2j that change the attenuation rate in the range from 0 dB to the minus side in accordance with the adjustment signals SF21 to SF2j from the frequency characteristic correction unit 11. Variable gain type frequency discriminating means, an adder ADD2, an inter-channel attenuator ATG2 for changing the attenuation rate in the range from 0 dB to a negative side according to the adjustment signal SG2 from the inter-channel level correction unit 12, and a phase A delay circuit DLY2 for changing the delay time in accordance with the adjustment signal SDL2 from the characteristic correction unit 13 is provided.

  The system circuit CQT3 of the third channel (x = 3) includes j band-pass filters BPF31 to BPF3j set to the above center frequencies f1 to fj, and the interband between the switch elements SW31 and SW32. J variable gain type frequency discriminating means constituted by attenuators ATF31 to ATF3j are provided, and an adder ADD3, an interchannel attenuator ATG3, and a delay circuit DLY3 are further provided. Similarly to the system circuit CQT1, the adjustment signals SF31 to SF3j from the frequency characteristic correction unit 11, the adjustment signal SG3 from the interchannel level correction unit 12, and the adjustment signal SDL3 from the phase characteristic correction unit 13 The attenuators ATF31 to ATF3j, the interchannel attenuator ATG3, and the delay circuit DLY3 are adjusted.

  The system circuit CQT4 of the fourth channel (x = 4) includes j band-pass filters BPF41 to BPF4j set to the center frequencies f1 to fj, and the band following the switch elements SW41 and SW42. J variable gain type frequency discriminating means constituted by inter-attenuators ATF41 to ATF4j are provided, and further, an adder ADD4, an inter-channel attenuator ATG4 and a delay circuit DLY4 are provided. Similarly to the system circuit CQT1, the adjustment signal SF41 to SF4j from the frequency characteristic correction unit 11, the adjustment signal SG4 from the interchannel level correction unit 12, and the adjustment signal SDL4 from the phase characteristic correction unit 13 The attenuators ATF41 to ATF4j, the interchannel attenuator ATG4, and the delay circuit DLY4 are adjusted.

  The system circuit CQT5 of the fifth channel (x = 5) includes j band-pass filters BPF51 to BPF5j set to the center frequencies f1 to fj, and the band following the switch elements SW51 and SW52. J variable gain type frequency discriminating means constituted by inter-attenuators ATF51 to ATF5j are provided, and further, an adder ADD5, an inter-channel attenuator ATG5 and a delay circuit DLY5 are provided. Similarly to the system circuit CQT1, the adjustment signal SF51 to SF5j from the frequency characteristic correction unit 11, the adjustment signal SG5 from the interchannel level correction unit 12, and the adjustment signal SDL5 from the phase characteristic correction unit 13 The attenuators ATF51 to ATF5j, the interchannel attenuator ATG5, and the delay circuit DLY5 are adjusted.

However, the system circuit CQTk of the sixth subwoofer channel (x = k) is i (i <j) that passes only the low frequency (frequency of about 0.2 KHz or less) shown in FIG. ) Band-pass filters BPFk1 to BPFki and inter-band attenuators ATFk1 to ATFki are connected in parallel following the switch elements SWk1 and SWk2, and the adder ADDk adds the outputs of the attenuators ATFk1 to ATFki, and the output of the addition result is the channel. The output DWF of the delay circuit DLYk is supplied to the D / A converter 4WF in FIG. 1 through the intermediate attenuator ATGk and the delay circuit DLYk.
The band-pass filters BPFk1 to BPFki and the interband attenuators ATFk1 to ATFki constitute i variable gain type frequency discriminating means.

  Next, in FIG. 3, when the frequency characteristic correction unit 11 individually rings each speaker 6FL, 6FR, 6C, 6RL, 6RR, 6WF by the noise signal (pink noise) DN output from the noise generator 3. Each sound collection data DM obtained is input, and the level of the reproduced sound of each speaker at the listening position RV is calculated based on the sound collection data DM. Based on the calculation results, adjustment signals SF11 to SF1j, SF21 to SF2j, to SFk1 to SFki are generated, and the attenuation factors of the interband attenuators ATF11 to ATF1j, ATF21 to ATF2j, to ATFk1 to ATFki are individually set. Automatically correct.

  By the correction of the attenuation rate by the frequency characteristic correction unit 11, gain correction is performed for each pass frequency of the bandpass filters BPF11 to BPFki provided in the system circuits CQT1 to CQTk for each channel.

  That is, the frequency characteristic correction unit 11 adjusts the level of each signal output from the bandpass filters BPF11 to BPFki by performing gain correction of the interband attenuators ATF11 to ATFki as the level adjustment means in the transmission path. The transmission line level correction means for setting the frequency characteristics.

  The channel-to-channel level correction unit 12 obtains each collection obtained when the full-band speakers 6FL, 6FR, 6C, 6RL, and 6RR are individually sounded by the noise signal (pink noise) DN output from the noise generator 3. The sound data DM is input, and the level of the reproduced sound of each speaker at the listening position RV is calculated based on the sound collection data DM. Based on the calculation result, adjustment signals SG1 to SG5 are generated, and the attenuation rates of the interchannel attenuators ATG1 to ATG5 are automatically corrected by the adjustment signals SG1 to SG5.

The level adjustment (gain adjustment) between the system circuits CQT1 to CQT5 of the first to fifth channels is performed by the attenuation rate correction of the interchannel level correction unit 12.
That is, the inter-channel level correction unit 12 is an inter-transmission path level correction unit that corrects the level of the audio signal transferred for each channel (signal transmission path) between the channels.

  However, the interchannel level correction unit 12 does not adjust the attenuation factor of the interchannel attenuator ATGk provided in the subwoofer channel system circuit CQTk, and the flattening correction unit 14 adjusts the attenuation factor of the interchannel attenuator ATGk. It has become.

  The phase characteristic correction unit 13 supplies a noise signal (non-correlated noise) DN output from the noise generator 3 to the system circuits CQT1 to CQTk of each channel, whereby each speaker 6FL, 6FR, 6C, 6RL, 6RR, 6WF is supplied. The phase characteristics of each channel are measured based on the respective sound collection data DM obtained when the sound is sounded individually, and the phase characteristics of the sound field space are corrected based on the measurement result.

  More specifically, the speakers 6FL, 6FR, 6C, 6RL, 6RR, and 6WF of each channel are sounded for each period T by the noise signal DN, and the sound collection data DM1, DM2, DM3, and DM4 of each channel that are generated thereby. , DM5, DMk are cross-correlated. Here, the cross-correlation between the sound collection data DM2 and DM1, the cross-correlation between the sound collection data DM3 and DM1, and then the cross-correlation between the sound collection data DMk and DM1 are calculated, and the peak interval (position) of each correlation value is calculated. (Phase difference) is defined as delay times τ2 to τk in the respective system circuits CQT2 to CQTk. That is, the delay times τ2 to τk of the remaining system circuits CQT2 to CQTk are obtained using the phase of the sound collection data DM1 obtained from the system circuit CQT1 as a reference (ie, phase difference 0, τ1 = 0). Based on the measurement results of these delay times τ1 to τk, adjustment signals SDL1 to SDLk are generated, and the delay times of the delay circuits DLY1 to DLYk are automatically adjusted by these adjustment signals SDL1 to SDLk, thereby Correct the phase characteristics. In this embodiment, uncorrelated noise is used to correct the phase characteristics, but pink noise or other noise signals may be used.

The flattening correction unit 14, after the adjustment by the frequency characteristic correction unit 11, the interchannel level correction unit 12, and the phase characteristic correction unit 13, is completed, the interchannel attenuator ATGk in the system circuit CQTk that is not adjusted by the interchannel level correction unit 12. Adjust the attenuation factor.
That is, as shown in FIG. 4, the flattening correction unit 14 includes a middle / high frequency processing unit 15a, a low frequency processing unit 15b, a subwoofer low frequency processing unit 15c, and a calculation unit 15d.

  The mid-high range processing unit 15a is configured to turn off the low-frequency bandpass filters BPF11 to BPF1i, BPF21 to BPF2i, BPF31 to BPF3i, BPF41 to BPF4i, and BPF51 to BPF5i provided in the system circuits CQT1 to CQT5. When all the band-type speakers 6FL, 6FR, 6C, 6RL, and 6RR are simultaneously ringed on the basis of the noise signal (uncorrelated noise) DN output from the noise generator 3 with the band-pass filter turned on. The spectrum average level PMH of the reproduced sound in the middle / high range is measured from the collected sound data DM (hereinafter referred to as the middle / high range collected data DMH).

  The low-frequency processing unit 15b conducts the low-frequency bandpass filters BPF11 to BPF1i, BPF21 to BPF2i, BPF31 to BPF3i, BPF41 to BPF4i, and BPF51 to BPF5i provided in the system circuits CQT1 to CQT5. When all band type speakers 6FL, 6FR, 6C, 6RL, and 6RR are simultaneously ringed based on a noise signal (uncorrelated noise) DN output from the noise generator 3 with the bandpass filter being non-conductive. The spectrum average level PL of the reproduced sound in the low range is measured from the collected sound data DM (hereinafter referred to as the low range collected data DL).

  The subwoofer low-frequency processing unit 15c sets all the bandpass filters BPFk1 to BPFki included in the subwoofer channel system circuit CQTk to the conductive state, and based on the noise signal (pink noise) DN output from the noise generator 3. From the sound collection data DM (hereinafter referred to as subwoofer sound collection data DWFL) obtained when the speaker 6WF dedicated to low frequency reproduction is sounded, the spectrum average level PWFL of the low sound reproduced only by the speaker 6WF is measured.

  The calculation unit 15d performs a predetermined calculation process, which will be described in detail later, on the basis of the above-described middle and high band average spectrum levels PMH and low band average spectrum levels PL and PWFL, thereby all the speakers 6FL, When 6FR, 6C, 6RL, 6RR, and 6WF are simultaneously sounded, an adjustment signal SGk for flattening the frequency characteristics of the reproduced sound at the listening position RV over the entire audio frequency band is generated.

  That is, as shown in the frequency characteristic diagram of FIG. 6, the all-band type speakers 6FL, 6FR, 6C, 6RL, and 6RR have a reproduction capability of not only the middle and high frequencies but also the low frequencies, so these speakers are used. When the 6FL, 6FR, 6C, 6RL, 6RR and the low frequency dedicated speaker 6WF are sounded, for example, the low frequency sound reproduced by the speakers 6FL, 6FR, 6C, 6RL, 6RR and the low frequency reproduced by the speaker 6WF In some cases, the spectrum average level with the sound becomes higher than the spectrum average level of the reproduced sound in the middle and high range, which causes a problem of being irritating or uncomfortable. Therefore, the calculation unit 15d adjusts the attenuation rate of the inter-channel attenuator ATGk by using the adjustment signal SGk so that the spectrum average level of the low frequency sound and the spectrum average level of the middle and high frequency band are flattened.

Therefore, the flattening correction unit 14 is an inter-transmission path level correction unit that corrects the level of the audio signal transferred for each channel (signal transmission path) together with the inter-channel level correction unit 12 between the channels.
Although the configuration of the automatic sound field correction system has been described, more detailed functions will be described in detail in the operation description.

  Next, the operation of the automatic sound field correction system having such a configuration will be described with reference to the flowcharts shown in FIGS.

  For example, as shown in FIG. 7, the listener arranges a plurality of speakers 6FL to 6WF in the listening room 7 and connects them to the audio system, and then a remote controller (not shown) provided in the audio system. Is operated to start sound field correction, the system controller MPU operates the automatic sound field correction system in accordance with this instruction.

  First, an outline of the operation of the automatic sound field correction system will be described with reference to FIG. In the frequency characteristic correction process in step S10, the frequency characteristic correction unit 11 adjusts the attenuation rate of all the interband attenuators ATF11 to ATFkj provided in the system circuits CQT1, CQT2, CQT3, CQT4, CQT5, and CQTk. Processing is performed.

  In the next inter-channel level correction process in step S20, the inter-channel level correction unit 12 adjusts the attenuation rate of the inter-channel attenuators ATG1 to ATG5 provided in the system circuits CQT1, CQT2, CQT3, CQT4, and CQT5. Processing is performed. That is, in step S20, the adjustment of the interchannel attenuator ATGk provided in the subwoofer channel system circuit CQTk is not performed.

  In the phase characteristic correction process in the next step S30, the phase characteristic correction unit 13 adjusts the delay times of all the delay circuits DLY1 to DLYk provided in the system circuits CQT1, CQT2, CQT3, CQT4, CQT5, and CQTk. Is performed. That is, processing for correcting the phase characteristics of reproduced sound reproduced by all the speakers 6FL to 6WF is performed.

  In the next flattening correction process in step S40, the flattening correction unit 14 performs a process for flattening the frequency characteristics of the reproduced sound at the listening position RV over the entire audio frequency band.

  As described above, the automatic sound field correction system performs sound field correction by sequentially performing correction processes roughly classified into four stages.

  Next, the processes of steps S10 to S40 will be described in order. First, the frequency characteristic correction process in step S10 will be described in detail. The process of step S10 is performed according to the detailed flow shown in FIG.

  In step S100, initialization processing is performed, and the attenuation rates of all the interband attenuators ATF11 to ATFki and interchannel attenuators ATG1 to ATGk of the system circuits CQT1, CQT2, CQT3, CQT4, CQT5, and CQTk shown in FIG. Set. Further, the delay times of all the delay circuits DLY1 to DLYk are set to 0, and the amplification factors of the amplifiers 5FL to 5WF shown in FIG. 1 are made equal.

  Further, the switch elements SW12, SW22, SW32, SW42, SW52, and SWk2 are turned off (non-conducting), whereby the input from the sound source 1 is cut off and the switch element SWN is turned on (conducting). Thus, the noise signal (pink noise) DN generated by the noise generator 3 is set to be supplied to each system circuit CQT1, CQT2, CQT3, CQT4, CQT5, CQTk.

  In step S102, flag data n = 0 is set in a flag register (not shown) built in the system controller MPU.

  Next, a sound field characteristic measurement process is performed in step S104. In this step S104, the switch elements SW11, SW21, SW32, SW41, SW51, SWk1 are exclusively turned on for each period of the predetermined period T, thereby supplying the noise signal DN to the system circuits CQT1 to CQTk in order, and further the noise. The band-pass filter of the system circuit to which the signal DN is supplied is made conductive in order from the low-frequency side to the middle-high frequency side.

  As a result, the noise signal DN frequency-divided by the bandpass filters BPF11 to BPF1j of the system circuit CQT1 is sequentially supplied to the speaker 6FL, whereby the microphone 8 collects the frequency-divided noise sound generated at the listening position RV. At the same time, the D / A converter 10 supplies the sound collection data DM (hereinafter referred to as DM11 to DM1j) to the frequency characteristic correction unit 11, and the frequency characteristic correction unit 11 further provides the sound collection data DM11 to DM11. DM1j is stored in a predetermined storage unit (not shown).

  Similarly, the noise signal DN frequency-divided through the remaining system circuits CQT2 to CQTk is supplied to the speakers 6FR to 6WF, and the resulting sound collection data DM (hereinafter referred to as DM21 to DM2j, DM31 to DM3j) for each channel. , DM41 to DM4j, DM51 to DM5j, DMk1 to DMki) are stored in a predetermined storage unit (not shown).

  Thus, by performing the sound field characteristic measurement processing, the frequency characteristic correction unit 11 stores the sound collection data [DAxJ] represented by the matrix of the following equation (1). Note that the suffix x in [DAxJ] indicates the channel number (1 ≦ x ≦ k), and the suffix J indicates the order of the center frequencies f1 to fj from the low range to the mid-high range (1 ≦ J ≦ j).

Further, in step S104, the sound collection data [DAxJ] and a predetermined threshold value THDCH are compared for each channel, and the sizes of the speakers 6FL to 6WF of each channel are determined based on the comparison result. That is, since the sound pressure of the reproduced sound from the speaker changes according to the speaker size, the speaker size of each channel is determined here.

  As specific determination means, the average value of the sound collection data DM11 to DM1j of the first channel in the above formula (1) is compared with the threshold value THDCH. When the average value is smaller than the threshold value THDCH, the speaker 6FL is determined as a small speaker, and when the average value is larger than the threshold value THDCH, the speaker 6FL is determined as a large speaker. The remaining speakers 6FR, 6FR, 6C, 6RL, 6RR, and 6WF are similarly determined.

  Then, the processing of steps S106 to S124 described below is not performed for the channel to which the speaker determined to be small is connected, and the processing of steps S106 to S124 is performed only for the channel to which the speaker determined to be large is connected. Do.

  In order to make the explanation easy to understand, it is assumed that the speakers 6FL, 6FR, 6FR, 6C, 6RL, 6RR, and 6WF are all large speakers, and the processing in steps S106 to S124 will be described.

  Next, in step S106, the target curve data [TGxJ] preset in the audio system by the listener is set in the frequency characteristic correction unit 11. Here, the target curve refers to the frequency characteristics of the reproduced sound that the listener likes. In this audio system, in addition to the target curve for generating the reproduced sound having the frequency characteristics suitable for classical music, rock music and Various target curve data [TGxJ] for generating a reproduction sound having frequency characteristics suitable for pops, vocals, etc. is stored in the system controller MPU. These target curve data [TGxJ] are composed of the same number of data sets as the interband attenuators ATF11 to ATFki, as shown by the matrix of the following equation (2), and can be selected independently for each channel. Yes.

When the listener operates a predetermined operation button of the remote controller, these target curves can be arbitrarily selected, and the system controller MPU sets the selected target curve data [TGxJ] in the frequency characteristic correction unit 11.
However, when the listener instructs the sound field correction without selecting the target curve, all the data TG11 to TGki are set to a predetermined value, for example, 1.

  Next, in step S108, the frequency characteristic correction unit 11 sets the number of the first channel (x = 1) and the order of the first center frequency (J = 1), and then repeats the processing of steps S110 to S114. Thus, adjustment values F0 (1,1) to F0 (1, j) for adjusting the interband attenuators ATF11 to ATF1j are calculated.

  That is, the flag data n is set to 0, the variable x representing the channel is set to 1, and the sound collection data [DAxJ] shown in the above formulas (1) and (2) is changed while the variable J is changed from 1 to j in steps S112 and S114. By applying the first row data DM11 to DM1j in the first row and the data TG11 to TG1j in the first row in the target curve data [TGAxJ] to the following equation (3), the interband attenuator corresponding to the first channel is applied. The adjustment values F0 (1,1) to F0 (1, j) of ATF11 to ATF1j are calculated. However, when the value TGxJ / DMxJ calculated by the expression (3) becomes a calculation error of a value smaller than a predetermined threshold value THD, the value TGxJ / DMxJ is forcibly set to 0 to improve the adjustment accuracy. I'm going to plan.

  Next, when it is determined in step S112 that all the adjustment values F0 (1,1) to F0 (1, j) of the interband attenuators ATF11 to ATF1j of the first channel have been calculated, the process proceeds to step S116, where It is determined whether the adjustment values of all the interband attenuators up to the sixth channel (x = 2 to k) have been calculated. If not, in step S118, variable x is incremented by 1 and variable j is set to 1, and the processing from step S110 is repeated. When calculation of the adjustment values of all the interband attenuators is completed, the process proceeds to step S120.

  Thereby, adjustment values [F0xJ] of all the interband attenuators ATF11 to ATF1j represented by the matrix of the following equation (4) are obtained.

  Next, in step S120, the adjustment value [F0xJ] is normalized by performing an operation represented by the matrix of the following equation (5), and the obtained normalized adjustment value [FN0xJ] is set as new target curve data [TGxJ]. ] = [FN0xJ]. That is, the target curve data [TGxJ] in the above equation (2) is replaced with the normalized adjustment value [FN0xJ].

  The values F01max to F0kmax with the suffix max in the equation (5) are the maximum adjustment values in the channels x = 1 to k when the flag data n is n = 1.

  Next, in step S122, it is determined whether or not the flag data n is 1. If NO (NO), the flag data n is set to 1 in step S124, and then the processing from step S104 is repeated.

  In this manner, the processing from step S104 is repeated, and if it is determined in step S122 that the flag data n is 1, the process proceeds to step S126. Here, when the processing from step S104 is repeated, the flag data is set to n = 1, and the calculations of the above formulas (1) to (5) are performed again, and the next corresponding to the above formula (5) is performed. The normalized adjustment value [FN1xJ] of Expression (6) is obtained.

  Next, in step S126, by multiplying the values of the respective matrices of the normalized adjustment values [FN0xJ] and [FN1xJ], all the interband attenuators ATF11 to ATF1j of the system circuits CQT1 to CQTk shown in Expression (7) are obtained. , To obtain adjustment data [SFxJ] for adjusting the attenuation rate of ATFk1 to ATFki.

  That is, the values F0 (1,1) / F01max and F1 (1,1) / F11max in the first row and the first column of the normalized adjustment values [FN0xJ] and [FN1xJ] shown in the equations (5) and (6). Is multiplied by the value SF11 of the first row and the first column of the matrix of the equation (7), and the values F0 (2,1) / F02max and F1 (2,1) of the second row and the first column are obtained. By multiplying / F12max, the value SF21 of the second row and the first column of the equation (7) is obtained, and the same calculation is performed thereafter to adjust the attenuation factor represented by the matrix of the equation (7). The adjustment data [SFxJ] is obtained.

  Then, after adjusting the attenuation rate of the interband attenuators ATF11 to ATF1j, to ATFk1 to ATFki by the adjustment signals SF11 to SF1j, to, SFk1 to SFki based on the adjustment data [SFxJ], the process proceeds to step S20 in FIG. .

  Further, in the sound field characteristic measurement process in step S104 described above, when a channel to which a small speaker is connected is determined, the attenuation factor of the interband attenuator provided in the channel is adjusted to 0 dB to increase the channel. The attenuation factor of the interband attenuator of the channel to which the speaker is connected is adjusted based on the adjustment data [SFxJ].

  If it is determined in step S104 that all the speakers 6FL, 6FR, 6FR, 6C, 6RL, 6RR, and 6WF are all small speakers, the process from step S104 is performed without performing the processes in steps S106 to S124. The process directly proceeds to step S126. In step S126, the attenuation factor of the interband attenuator of all channels is adjusted to 0 dB.

  As described above, the frequency characteristic for each channel is corrected by adjusting the attenuation rate of the interband attenuators ATF11 to ATFki by the frequency characteristic correction unit 11, and the frequency characteristic of the sound field space is optimized.

  Further, in the sound field characteristic measurement processing in step S104, each speaker 6FL, 6FR, 6C, 6RL, 6RR, 6WF is caused to ring in a time-sharing manner by pink noise, so that a sound field space is generated based on the actual audio signal. It is possible to detect the frequency characteristics and reproduction capability (output power) of each speaker under almost the same conditions as in the past. Therefore, it is possible to comprehensively correct the frequency characteristics in consideration of the frequency characteristics and reproduction ability of each speaker.

  Next, the inter-channel level correction process in step S20 is performed according to the flow shown in FIG.

First, initialization processing in step S200 is performed, and the switch elements SW11 to SW52 are switched to enable the input of the noise signal DN from the noise generator 3. However, the switch elements SWk1 and SWk2 of the subwoofer channel are turned off. Further, the attenuation factor of the inter-channel attenuators ATG1 to ATGk is set to 0 dB. Further, the delay times of all delay circuits DLY1 to DLY5 are set to zero. Furthermore, the amplification factors of the amplifiers 5FL to 5WF shown in FIG.
Further, the attenuation factors of the interband annators ATF11 to ATF1j, ATF21 to ATF2j,..., ATFk1 to ATFki are set to the fixed state as adjusted in the frequency characteristic correction process.

  Next, in step S202, after setting the variable x representing the channel number to 1, the sound field characteristic measurement processing in step S204 is performed, and further, until the sound field characteristic measurement for the first to fifth channels is completed. The processes in steps S204 to S208 are repeated.

  Here, the band pass filters BPF11 to BPF1j,..., BPF51 to BPF5j are always kept on (conductive), and the switch elements SW11, SW21, SW31, SW41, and SW51 are exclusively turned on by a predetermined period T, A noise signal (pink noise) DN is sequentially supplied to the system circuits CQT1 to CQT5 (steps S206 and S208).

  Through this repeated processing, the microphone 8 collects the reproduced sounds reproduced by the speakers 6FL, 6FR, 6C, 6RL, and 6RR, and the collected sound data DM (= DM1) for each of the first to fifth channels obtained thereby. To DM5) are stored in a memory unit (not shown) in the inter-channel level adjusting unit 13. That is, the sound collection data [DBx] represented by the matrix of the following equation (8) is stored.

  Next, when the measurement of the sound field characteristics of the first to fifth channels is completed, the process proceeds to step S210, where one piece of minimum sound collection data is extracted from the sound collection data DM1 to DM5, and the extracted data Is the target data TGCH for level adjustment between channels.

  Next, in step S212, the matrix of the above equation (8) is normalized with the target data TGCH for adjusting the level between channels, thereby adjusting the attenuation rate of each channel attenuator ATG1 to ATG5 shown in the following equation (9). After obtaining the value [SGx], in step S214, the attenuation factors of the interchannel attenuators ATG1 to ATG5 are adjusted by the adjustment signals SG1 to SG5 based on the attenuation factor adjustment value [SGx].

  With the above processing, level adjustment is completed only between the first to fifth channels to which the full-band speakers are connected, excluding the subwoofer channel, and then the process proceeds to step S30 in FIG.

  As described above, the level characteristic of each channel is optimized by correcting the attenuation rate of the inter-channel attenuators ATG1 to ATGk by the inter-channel level correction unit 12, and the level of the reproduced sound of each speaker at the listening position RV is optimized. To do.

  Further, in the sound field characteristic measurement processing in step S204, each speaker 6FL, 6FR, 6C, 6RL, 6RR is caused to ring in a time-sharing manner, and the reproduced sound generated thereby is collected. ) Can be detected. For this reason, it is possible to make a comprehensive optimization considering the reproduction ability of each speaker.

  Next, the phase characteristic correction process of step S30 is performed according to the flow shown in FIG.

  First, initialization processing in step S300 is performed, and the switch elements SW11 to SWk2 are switched to enable the input of the noise signal (uncorrelated noise) DN output from the noise generator 3. Further, the interband attenuators ATF11 to ATFki and the interchannel attenuators ATG1 to ATGk are fixed at the already adjusted attenuation rates, and the delay times of the delay circuits DLY1 to DLYk are set to zero. Furthermore, the amplification factors of the amplifiers 5FL to 5WF shown in FIG.

  Next, in step S302, the variable x representing the channel number is set to 1 and the variable AVG is set to 0, and then the sound field characteristic measurement process for measuring the delay time in step S304 is performed. Steps S304 to S308 are repeated until the sound field characteristic measurement for the channels is completed.

  Here, the switch elements SW11, SW21, SW31, SW41, and SWk1 are exclusively turned on by a predetermined period T, and a noise signal (uncorrelated noise) DN is supplied to the system circuits CQT1 to CQTk for each period T.

  By repeating this process, a continuous noise signal DN is supplied to each speaker 6FL, 6FR, 6C, 6RL, 6RR, 6WF for each period T, and each reproduced sound of the noise signal DN reproduced for each period T is reproduced. Macrolophone 8 collects sound. Further, the phase characteristic correction unit 13 inputs each sound collection data DM (hereinafter referred to as DM1, DM2, DM3, DM4, DM5, DMk) output from the A / D converter 10 for each period T. Since the high-speed sampling is performed by the A / D converter 10 every period T, these collected sound data DM1, DM2, DM3, DM4, DM5, DMk are each a plurality of sampling data.

  When this measurement ends, the process proceeds to step S310, and the phase characteristics of each channel are calculated. Here, the cross-correlation operation is performed on the collected sound data DM2 and DM1, and the peak interval (phase difference) of the correlation value obtained thereby is set as the delay time τ2 in the system circuit CQT2. Further, the cross-correlation with the sound collection data DM1 is calculated for each of the remaining sound collection data DM3 to DMk, and the peak interval (phase difference) of each correlation value obtained thereby is calculated as the delay time in the system circuits CQT3 to CQTk. Let τ3 to τk. That is, the delay times τ2 to τk of the remaining system circuits CQT2 to CQTk are calculated using the phase of the sound collection data DM1 obtained from the system circuit CQT1 as a reference (that is, a phase difference of 0).

  Next, the process proceeds to step S312, and 1 is added to the variable AVG. Then, in step S314, it is determined whether or not the variable AVG has reached a predetermined value AVRAGE. If not, the process from step S304 is repeated.

  Here, the predetermined value AVRAGE is a constant indicating the number of repetitions of steps S304 to S312. In this embodiment, AVRAGE = 4.

  By repeating the measurement process four times in this way, four delay times τ1 to τk of each of the system circuits CQT1 to CQTk are obtained, respectively. Next, in step S316, the average value of each of the four delay times τ1 to τk. τ1 ′ to τk ′ are obtained, and the average values τ1 ′ to τk ′ are set as delay times of the respective circuit circuits CQT1 to CQTk. The delay times are SDL1 to SDLk.

  In step S318, the delay times of the delay circuits DLY1 to DLYk are adjusted based on the adjustment signals SDL1 to SDLk corresponding to the delay times τ1 'to τk', and the phase characteristic correction process is completed.

  As described above, in the phase characteristic correction processing, the noise signal for measuring the delay time is supplied to each speaker through the system circuits CQT1 to CQTk and caused to ring, and the phase characteristic is obtained from the sound collection result of the reproduced sound generated thereby. Therefore, instead of simply adjusting (correcting) the delay times of the delay circuits DLY1 to DLYk only from the propagation delay time of the reproduced sound, comprehensive optimization considering the reproduction capability of each speaker and the characteristics of the system circuits CQT1 to CQTk Is possible.

  Next, when the phase characteristic correction process is completed, the process proceeds to the flattening correction process in step S40 in FIG. The process of step S40 is performed according to the flow shown in FIG. First, in step S400, the switch elements SW11 to SWk1 are switched so that the noise signal (uncorrelated noise) DN output from the noise generator 3 can be input. Further, the amplification factors of the amplifiers 5FL to 5WF are made equal.

  Next, in step S402, the interband attenuators ATF11 to ATFki, the interchannel attenuators ATG1 to ATG5, and the delay circuits DLY1 to DLYk are fixed in the already adjusted state. However, in step S404, the attenuation factor of the inter-channel attenuator ATGk in the system circuit CQTk is set to 0 dB.

  Next, in step S406, a noise signal (non-correlated noise) DN is simultaneously supplied to the system circuits CQT1 to CQT5 except for the system circuit CQTk. Here, among the interband attenuators ATF11 to ATF1j,..., ATF51 to ATF5j in the system circuits CQT1 to CQT5, the interband attenuators ATF11 to ATF1i,. The noise signal DN is supplied.

  As a result, the full-band speakers 6FL, 6FR, 6C, 6RL, and 6RR are simultaneously sounded by the mid-high frequency noise signal DN, and the mid-high frequency sound collection data DMH (see FIG. 4) obtained thereby is mid-high frequency processing unit 15a. Further, based on this mid-high range sound collection data DMH, the spectrum average level PMH of the reproduced sound in the mid-high range by the speakers 6FL, 6FR, 6C, 6RL, 6RR is calculated.

  Next, in step S408, a noise signal (uncorrelated noise) DN is simultaneously supplied to the system circuits CQT1 to CQT5 except for the system circuit CQTk. Here, among the interband attenuators ATF11 to ATF1j,..., ATF51 to ATF5j in the system circuits CQT1 to CQT5, the interband attenuators ATF11 to ATF1i,. The interband attenuator is set to an off (non-conducting) state and supplies the noise signal DN.

  As a result, the full-band speakers 6FL, 6FR, 6C, 6RL, and 6RR are simultaneously sounded by the low-frequency noise signal DN, and the low-frequency sound collection data DL (see FIG. 4) obtained thereby is transmitted to the low-frequency processing unit 15b. Further, based on the low-frequency sound collection data DL, the spectrum average level PL of low-frequency reproduced sound by the speakers 6FL, 6FR, 6C, 6RL, and 6RR is calculated.

  Next, in step S410, a noise signal (pink noise) DN is supplied only to the system circuit CQTk. Here, among the ATF11 to ATF1j,..., ATF51 to ATF5j, the interband attenuators ATF11 to ATF1i,. And the noise signal DN is supplied.

  As a result, only the low-frequency reproduction speaker 6WF is activated by the noise signal DN, and the subwoofer low-frequency processing unit 15c inputs the subwoofer sound collection data DWFL (see FIG. 4) obtained thereby. Based on the data DWFL, the spectrum average level PWFL of the low frequency reproduced sound by the speaker 6WF is calculated.

  Next, in step S412, the calculation unit 14 performs an operation represented by the following equation (10) to obtain an adjustment signal SGk for adjusting the attenuation factor of the inter-channel attenuator ATGk of the system circuit CQTk.

  That is, by performing the calculation of the above expression 10, when audio reproduction is performed with all the speakers 6FL, 6FR, 6C, 6RL, 6RR, and 6WF, the frequency characteristics of the reproduced sound in the sound field space are made flat. An adjustment signal SGk is obtained.

  More specifically, among the reproduced sounds simultaneously reproduced by the all-band speakers 6FL, 6FR, 6C, 6RL, and 6RR, the level of the reproduced sound of the low frequency and the reproduced sound reproduced by the subwoofer 6WF dedicated to the low frequency range. The target characteristic (characteristic represented by the target curve data) is the sum of the level and the level of the reproduced sound in the middle and high frequencies of the reproduced sound simultaneously reproduced by all-band speakers 6FL, 6FR, 6C, 6RL, 6RR. An adjustment signal SGk for adjusting the attenuation factor of the inter-channel attenuator ATGk is obtained so as to be equal to the ratio.

  The coefficient TGMH in the above equation (10) is the target curve data selected by the listener from the target curve data [TGxJ] shown in the above equation (2) or the default target when the listener does not select the target curve data. Of the curve data, it is the average value of the target curve data corresponding to the mid-high range. The coefficient TGL is an average value of the target curve data corresponding to the low range.

  Next, in step S414, the attenuation rate of the inter-channel attenuator ATGk is adjusted by the adjustment signal SGk, and the automatic sound field correction process is completed.

  As described above, when level correction between channels is finally performed by the flattening correction unit 14, when audio reproduction is performed by all the speakers 6FL, 6FR, 6C, 6RL, 6RR, and 6WF, reproduction in the sound field space is performed. The frequency characteristic of sound can be made flat in the entire audio frequency band. For this reason, the conventional problem that the level of the low frequency shown in FIG. 6 becomes large can be solved.

  Further, in the sound field characteristic measurement processing in steps S404 to S410, each speaker 6FL, 6FR, 6C, 6RL, 6RR, 6WF is caused to ring in a time-sharing manner, and the reproduced sound generated thereby is collected. Capability (output power) can be detected. For this reason, it is possible to make a comprehensive optimization considering the reproduction ability of each speaker.

  Then, the switch element SWN is turned off, the switch elements SW11, SW21, SW31, SW41, SW51, SWk1 connected to the switch element are turned off, and the switch elements SW12, SW22, SW32, SW42, SW52, SWk2 are turned on. As a result, the audio signals SFL, SFR, SC, SRL, SRR, SWF from the sound source 1 are set to the input-ready state, and the audio system enters a normal audio playback state.

  As described above, according to the present embodiment, the frequency characteristic and phase characteristic of the sound field space are corrected in consideration of the characteristics of the audio system and the speaker, so that the sound field having an extremely high quality and presence is obtained. Space can be provided.

  Further, the problem that the level of reproduced sound at a certain frequency in the audio frequency band is increased or decreased, for example, the problem that the low frequency level shown in FIG. 6 is increased can be solved. In other words, since the frequency characteristics of the playback sound played back by each speaker is flattened over the entire audio frequency band, the playback sound of a certain frequency becomes strong, and an irritating sound can be heard or uncomfortable. This eliminates the problem and realizes a very high-quality and realistic sound field space.

  Further, by performing the sound field correction processing in the order of steps S10 to S40 shown in FIG. 8, it is possible to perform correction that realizes an extremely high-quality and realistic sound field space.

  In addition, since the sound field is corrected according to the target curve designated by the listener, it is possible to improve convenience.

  Also, when correcting frequency characteristics, correcting the level between channels, and flattening, pink noise similar to the frequency characteristics of the audio signal is used. ing.

  In the present embodiment, an automatic sound field correction system for a so-called 5.1-channel multi-channel audio system including the wide-band speakers 6FL to 6RR for five channels and the low-frequency speaker 6WF has been described. It is not limited to. The automatic sound field correction system of the present invention can be applied to a multi-channel audio system including a larger number of speakers than in the present embodiment, and can also be applied to an audio system including a smaller number of speakers than in the present embodiment.

  In other words, the present invention can be applied to an audio system including one or more speakers.

  Further, although the sound field correction in the audio system provided with the low-frequency reproduction dedicated speaker (subwoofer) 6WF has been described, the present invention is not limited to this. Even in an audio system including only a full-band speaker without a subwoofer, a high-quality and realistic sound field space can be provided. In this case, the flattening correction unit 14 may not be provided, and the characteristics of all channels may be corrected by the inter-channel level correction unit 12.

  In this embodiment, in step S412 shown in FIG. 12, the attenuation rate of the inter-channel attenuator ATGk is based on the level of the reproduced sound of the all-band speakers 6FL to 6RR, as is apparent from the equation (10). Is being optimized. That is, the denominator of the expression (10) is the product of the medium / high frequency target data TGMH and the variable PWFL corresponding to the spectrum average level of the reproduced sound of the low frequency dedicated speaker 6WF, so that the all-band type speakers 6FL˜ Based on 6RR playback sound level. However, the present invention is not limited to this, and the attenuation rate of the interchannel attenuators AT1 to AT5 may be optimized based on the level of the reproduced sound of the speaker 6WF dedicated to the low range.

  That is, in the present embodiment, the flattening correction unit 14 corrects the attenuation rate of the inter-channel attenuator ATGk. On the contrary, the level of the reproduced sound of the low-frequency speaker 6WF is measured, and the measurement result The attenuation rate of the interchannel attenuator ATGk may be set based on the above, and the attenuation rates of the interchannel attenuators ATG1 to ATG5 may be corrected based on the attenuation rate of the interchannel attenuator ATGk.

  Further, as described above, each of the system circuits CQT1 to CQTk shown in FIG. 2 is configured by connecting the bandpass filter, the interband attenuator, the adder, the interchannel attenuator, and the delay circuit in this order. It is shown as a typical example, and the present invention is not limited to such a configuration.

  For example, a delay circuit that is subordinately connected to the inter-channel attenuator may be disposed on the input side of the band-pass filter, or may be disposed on the input side of the inter-band attenuator. Further, the positions of the interchannel attenuator and the delay circuit may be interchanged. Further, both the inter-channel attenuator and the delay circuit may be arranged on the input side of the band pass filter.

  Unlike the conventional audio system in which the correction of the frequency characteristics and the correction of the phase characteristics are performed separately for each component, the present invention can be configured such that the positions of these components are appropriately changed. This is because the noise signal from the device is input from the input stage of the sound field correction system, and the frequency characteristic and phase characteristic of the entire sound field correction system are corrected comprehensively. As a result, the sound field correction system of the present invention can appropriately correct the frequency characteristics and phase characteristics of the entire audio system, and can also increase the degree of design freedom.

It is a block diagram which shows the structure of an audio system provided with the automatic sound field correction system which concerns on this embodiment. It is a block diagram which shows the structure of an automatic sound field correction system. It is a block diagram which shows the principal part structure of an automatic sound field correction system. It is a block diagram which further shows the principal part structure of an automatic sound field correction system. It is a figure which shows the frequency characteristic of a band pass filter. Illustration for explaining problems in the low frequency range of playback sound It is a figure which shows the example of arrangement | positioning of a speaker. It is a flowchart for demonstrating operation | movement of an automatic sound field correction system. It is a flowchart for demonstrating a frequency characteristic correction process. It is a flowchart for demonstrating the level correction process between channels. It is a flowchart for demonstrating a delay characteristic correction process. It is a flowchart for demonstrating flattening correction processing.

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 ... Sound source 2 ... Signal processing circuit 3 ... Noise generator 8 ... Microphone 9 ... Amplifier 10 ... A / D converter 11 ... Frequency characteristic correction part 12 ... Inter-channel level correction part 13 ... Delay characteristic correction part 14 ... Flattening correction Unit 15a ... Middle high frequency processing unit 15b ... Low frequency processing unit 15c ... Subwoofer low frequency processing unit 6FL, 6FR, 6C, 6RL, 6RR, 6WF ... Speakers CQT1-CQTk ... System circuits BPF11-BPFki ... Bandpass filters ATF11-ATFki ... Interband attenuators ADD1 to ADDk ... Adders ATG1 to ATGk ... Interchannel attenuators DLY1 to DLYk ... Delay circuits SW11 to SWk2, SWN ... Switch elements MPU ... System controller

Claims (4)

Each of the plurality of channels includes a signal processing system for performing signal processing on the audio signal and sound emitting means for reproducing sound based on the output of the signal processing system,
In the signal processing system of each channel,
Frequency dividing means that is a plurality of filters that discriminate and output the audio signal into a plurality of frequency bands;
An interband level adjusting means that is an attenuator for performing a plurality of gain adjustments provided in association with the plurality of filters;
Summing means for generating a sum signal by adding the signals discriminated by the frequency dividing means and gain-adjusted by the interband level adjusting means;
An interchannel level adjusting means for performing gain adjustment on the addition signal;
A sound field correction method in an audio system , comprising delay means for performing delay processing on a signal subjected to gain adjustment by the inter-channel level adjusting means and outputting the signal to a sound emitting means,
In the signal processing systems of all channels, the gains of the plurality of attenuators as the interband level adjusting means are made equal, the gains of the interchannel level adjusting means are made equal, the delay times of the delay means are made equal, and noise the supplied exclusively and sequentially to the signal processing system for each channel, and exclusively conducting each of the plurality of filters is a frequency division means for the noise is provided in the supplied signal processing system in order Discriminating the noise into the frequency band for each filter in which the noise is conducted in the order and exclusively, and reproduced by a speaker provided in a channel of a signal processing system that outputs the discriminated noise. that by detecting at the listening position noise sound off provided in the signal processing system of all channels A first detection step of detecting a noise sound of the total number fraction of filters,
The sound emission means for reproducing the noise sound having a sound pressure larger than the threshold by comparing the sound pressure of each noise sound detected in the first detection step with a predetermined threshold, and the noise having a sound pressure smaller than the threshold A plurality of attenuators , which are channel level adjustment means for channels provided with the sound emission means for reproducing the noise sound having the determined high sound pressure, and a predetermined target curve. gain adjustment, and by not more than one attenuator is the band between the level adjusting means of the channel sounding means reproduces the noise sound of the determination by the small sound pressure is provided to gain adjustment, the signal processing for each channel in accordance with A frequency characteristic correcting step for setting a frequency characteristic by the frequency dividing means and the interband level adjusting means in the system;
The frequency division means provided in the signal processing system of the selected channel for selecting the signal processing system of each channel for which the frequency characteristic is set in the frequency characteristic correction step exclusively and sequentially for each channel. By supplying noise to a plurality of filters at the same time , the frequency characteristics of the noise are obtained by the plurality of filters and a plurality of attenuators serving as interband level adjusting means provided in the signal processing system of the selected channel. A second detecting step of setting and detecting the sound pressure of the noise sound reproduced by the sound emitting means provided in the channel of the selected signal processing system at the listening position;
A plurality of attenuators as level adjustment means between channels provided in the signal processing system of each channel based on the minimum value of the sound pressure of the noise sound detected for each channel in the second detection step; An interchannel level correction process for adjusting the gain;
The signal processing system of each channel for which the frequency characteristic is set and the gain is adjusted in the frequency characteristic correction step and the inter-channel level correction step is selected exclusively and sequentially for each channel, and the channel of the selected channel is selected. supplying simultaneously noise for the plurality of filters is a frequency division means provided in the signal processing system, noise sound is reproduced based on the noise output from the signal processing system of channel that the selected to sound emission means A third detection step of detecting at the listening position;
Based on the phase difference of each noise sound detected for each channel in the third detection step, the delay time of the delay means provided in the signal processing system for each channel is determined from the sound emission means for each channel. A phase characteristic correction step for adjusting a delay time corresponding to the distance to the listening position;
A sound field correction method comprising: A signal processing system (hereinafter referred to as a “low frequency channel signal processing system”) that performs signal processing on a low frequency range of an audio signal and a sound emitting means that reproduces sound based on the output of the low frequency channel signal processing system; And a signal processing system (hereinafter referred to as “global channel signal processing system”) that performs signal processing on the entire range of the audio signal and a sound that is reproduced based on the output of the global channel signal processing system. Sound means is provided for each of the plurality of global channels,
The signal processing system of each global channel includes
A first frequency dividing means that is a plurality of filters for discriminating and outputting the entire audio signal into a plurality of frequency bands;
A first interband level adjusting means that is an attenuator that performs a plurality of gain adjustments provided in association with the plurality of filters;
First addition means for generating a first addition signal by adding the signals discriminated by the first frequency division means and gain-adjusted by the first interband level adjustment means;
First inter-channel level adjusting means for performing gain adjustment on the first addition signal;
First delay means for performing delay processing on the signal that has been gain-adjusted by the first inter-channel level adjusting means and outputting the signal to the sound emitting means,
The low-frequency channel signal processing system includes
A second frequency dividing means which is a plurality of filters for discriminating and outputting the low frequency range of the audio signal into a plurality of frequency bands;
A second interband level adjusting means that is an attenuator that performs a plurality of gain adjustments provided in association with a plurality of filters that are the second frequency dividing means;
Second addition means for generating a second addition signal by adding the signals discriminated by the second frequency dividing means and gain-adjusted by the second interband level adjusting means;
Second inter-channel level adjusting means for performing gain adjustment on the second addition signal;
A sound field correction method in an audio system , comprising: a second delay unit that performs a delay process on the signal that has been gain-adjusted by the second inter-channel level adjusting unit and outputs the signal to a sound emitting unit. There,
In all the all-band channel signal processing system and the low-frequency channel signal processing system, the gains of the plurality of attenuators as the first and second interband level adjusting means are made equal and the first and second interchannel levels are equalized. The gain of the adjusting means is made equal, the delay times of the first and second delay means are made equal, and noise is supplied exclusively and sequentially to the signal processing systems of the all-band channel and the low-band channel, For each filter in which the noise is supplied in order and exclusively by conducting each of the plurality of filters , which are frequency division means provided in the signal processing system supplied with noise, in order and exclusively. Discriminating into the frequency band of the signal and the signal reproduced by the sound emitting means provided in the channel of the signal processing system that outputs the discriminated noise. A first detection step of detecting noise sounds for the total number of filters provided in the signal processing systems of all channels by detecting noise at the listening position;
The sound emission means for reproducing the noise sound having a sound pressure larger than the threshold by comparing the sound pressure of each noise sound detected in the first detection step with a predetermined threshold, and the noise having a sound pressure smaller than the threshold Of the channel provided with the sound emitting means for reproducing the noise sound having the determined large sound pressure among the first and second inter-band level adjusting means . By adjusting the gain of the interband level adjusting means according to a predetermined target curve and not adjusting the gain of the interband level adjusting means of the channel provided with the sound emitting means for reproducing the noise sound of the determined small sound pressure A frequency characteristic correction step for setting frequency characteristics by the frequency dividing means and the interband level adjusting means in the signal processing system of each channel;
Select the frequency characteristic correcting process in the each entire channel signal processing system of which the frequency property has been set to the exclusive and sequentially for each channel, the first frequency division provided to the selected entire channel signal processing system By simultaneously supplying noise to a plurality of filters as means , the frequency characteristics of the noise are adjusted by the first interband level adjusting means provided in the plurality of filters and the selected global channel signal processing system. And a second attenuator configured to detect the sound pressure of a noise sound to be reproduced based on the noise output from the selected global channel signal processing system to the sound emission means of the channel at the listening position. A detection process;
A plurality of attenuators as first inter-channel level adjusting means provided in each of the global channel signal processing systems based on the minimum value of the sound pressure of the noise sound detected for each global channel in the second detection step An inter-channel level correction process for adjusting the gain of
Each frequency channel signal processing system in which the frequency characteristic is set and the gain is adjusted in the frequency characteristic correction step and the inter-channel level correction step , and the low frequency range in which the frequency characteristic is set in the frequency characteristic correction step select a channel signal processing system in the exclusive and one by one for each channel, and simultaneously supplies a noise to a plurality of filters is a frequency division means provided in the signal processing system of the channel that the selected, the A third detection step of detecting the sound pressure of the noise sound reproduced based on the noise output from the signal processing system of the selected channel to the sound output means of the channel at the listening position;
Based on the phase difference of each noise sound detected for each channel in the third detection step, the first delay means provided in each of the entire channel signal processing systems and the low frequency channel signal processing system and phase characteristic correcting step of adjusting the delay time of the second delay means provided, the delay time corresponding to the distance from sound emission unit of each channel to the listening position,
After the phase characteristic correcting process, it is supplied simultaneously noise to all of the all-channel signal processing system, the filter according to the low frequency of the plurality of filters is a respective first frequency dividing means to which the noise has been supplied The attenuator of the interband level adjusting means provided is non-conducting, and the attenuator of the interband level adjusting means provided in the filter relating to the middle / high range is made conductive, and is provided in all the global channel signal processing systems. The middle and high frequency noise sound reproduced simultaneously by the sound emitting means is detected at the listening position, and then the noise is simultaneously supplied to all the global channel signal processing systems, and each first frequency division to which the noise is supplied conducting an attenuator inter-band level adjusting means provided in the filter according to the low frequency of the plurality of filters is a means, a medium And an attenuator inter-band level adjusting means provided in the filter according to the frequency non-conductive, the first low-noise sound which is played simultaneously with sound means provided on the all of the all-channel signal processing system detected by the listening position as frequency noise sound, then the simultaneous supply noise to a plurality of filters is a second frequency dividing means in the low-frequency channel signal processing system, provided in the low-frequency channel signal processing system A fourth detection step of detecting a noise sound reproduced by the sound emitting means as a second low-frequency noise sound at the listening position;
The low-frequency spectrum of the second low-frequency noise sound with respect to the difference between the mid-high frequency spectrum of the middle-high frequency noise sound detected in the fourth detection step and the low-frequency spectrum of the first low-frequency noise sound Based on the ratio, by adjusting the gains of a plurality of attenuators which are second channel level adjusting means provided in the low frequency channel signal processing system , the sound emission means of the global channels and the low frequency channel A flattening correction step for flattening the frequency characteristics of the reproduced sound in the sound field space reproduced by the sound emitting means,
A sound field correction method comprising: In the frequency characteristic correction step, an average value of sound pressure for each channel is calculated based on a sound pressure of each noise sound detected in the first detection step, and an average value of sound pressure for each channel is calculated. Comparing the threshold value with the threshold value,
The sound field correction method according to claim 1 or 2, characterized by the above-mentioned. The audio system is a 5.1 channel multi-channel audio system;
The sound field correction method according to claim 1 or 2, characterized by the above-mentioned.

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