WO2015159901A1 - 放収音装置及び放収音方法 - Google Patents
放収音装置及び放収音方法 Download PDFInfo
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- WO2015159901A1 WO2015159901A1 PCT/JP2015/061520 JP2015061520W WO2015159901A1 WO 2015159901 A1 WO2015159901 A1 WO 2015159901A1 JP 2015061520 W JP2015061520 W JP 2015061520W WO 2015159901 A1 WO2015159901 A1 WO 2015159901A1
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M1/00—Substation equipment, e.g. for use by subscribers
- H04M1/60—Substation equipment, e.g. for use by subscribers including speech amplifiers
- H04M1/6033—Substation equipment, e.g. for use by subscribers including speech amplifiers for providing handsfree use or a loudspeaker mode in telephone sets
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M1/00—Substation equipment, e.g. for use by subscribers
- H04M1/60—Substation equipment, e.g. for use by subscribers including speech amplifiers
- H04M1/6016—Substation equipment, e.g. for use by subscribers including speech amplifiers in the receiver circuit
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M1/00—Substation equipment, e.g. for use by subscribers
- H04M1/60—Substation equipment, e.g. for use by subscribers including speech amplifiers
- H04M1/62—Constructional arrangements
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M9/00—Arrangements for interconnection not involving centralised switching
- H04M9/08—Two-way loud-speaking telephone systems with means for conditioning the signal, e.g. for suppressing echoes for one or both directions of traffic
- H04M9/082—Two-way loud-speaking telephone systems with means for conditioning the signal, e.g. for suppressing echoes for one or both directions of traffic using echo cancellers
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M2203/00—Aspects of automatic or semi-automatic exchanges
- H04M2203/50—Aspects of automatic or semi-automatic exchanges related to audio conference
- H04M2203/509—Microphone arrays
Definitions
- the present invention relates to a sound emission and collection device used for remote audio conferences and the like, and more particularly to suppression of reverberation of sound to be emitted.
- An audio conference system that transmits and receives audio by connecting bases via a network has been put into practical use.
- the acoustic characteristics of conference rooms used for conferences vary, and conferences may be held in rooms with very long reverberations. If the reverberation is long, the clarity of the sound emitted from the speaker decreases.
- Patent Document 1 An apparatus for suppressing the reverberation of the emitted sound has been proposed.
- the device of Patent Document 1 operates the key Kia corresponding to the participant Ma, so that the inverse filter coefficient Ga of the spatial transfer function Ha from the participant Ma to the microphone 31 is read from the ROM 41 and applied to the digital filter 34i.
- the supplied digital filter 34i performs inverse filter operation in real time to perform inverse filtering on the voice signal of the participant Ma. That is, in this apparatus, the spatial transfer functions from the seats of the participants Ma to Mn to the plurality of microphones 31 are measured in advance, and the inverse filter coefficients Ga to Gn of the transfer functions are stored in the ROM 41 in advance.
- test signal is played back in the presence of the participant. It is preferable to do this.
- An object of the present invention is to provide a sound emitting and collecting apparatus and a sound emitting and collecting method capable of suppressing reverberation using a sound collecting and echo canceling function for a conference without reproducing test audio in advance.
- a sound emission and collection device includes a speaker, a filter that processes a sound emission signal that is an audio signal supplied to the speaker, a plurality of microphones, and each of the plurality of microphones.
- a plurality of echo cancellers provided correspondingly, each canceling a return sound signal of the sound emitted by the speaker from the sound pickup signal of the corresponding microphone, and adaptive filter coefficients extracted from the plurality of echo cancellers
- a reverberation time estimation unit for estimating a reverberation time for each frequency band of the space where the speaker and the plurality of microphones exist based on the integrated adaptive filter coefficient, and the estimated Based on the reverberation time, a frequency band having a long reverberation time is identified from the sound emission signal, and the power of the specified frequency band is suppressed.
- the sound emission and collection method is a method in which a sound emission signal that is an audio signal supplied to a speaker is processed by a filter and collected by the plurality of microphones by a plurality of echo cancellers provided corresponding to each of the plurality of microphones. Canceling the return signal of the sound emitted by the speaker from the sound signal, integrating the adaptive filter coefficients extracted from the plurality of echo cancellers, and based on the integrated adaptive filter coefficients, the speaker and the plurality The reverberation time for each frequency band in the space where the microphone is present is estimated, and based on the estimated reverberation time, a frequency band having a long reverberation time is identified from the sound emission signal, and the power of the identified frequency band is determined. Is calculated and set in the filter.
- the present invention it is possible to appropriately suppress reverberation using a directional microphone suitable for a conference and using parameters of an echo canceller (such as filter coefficients of an adaptive filter).
- FIGS. 4A to 4C are diagrams illustrating an example in which individual microphones are grouped to form a sound collector.
- FIG. 1 is a diagram illustrating an example of an installation form of an audio conference system according to an embodiment of the present invention.
- the audio conference system 1 is installed on the conference table D in the conference room C.
- the audio conference system 1 has one communication device 10 and one or a plurality of sound collectors 11 (four in this embodiment).
- the communication device 10 has a speaker 26.
- the sound collector 11 includes a plurality of microphone elements 31.
- the communication device 10 and the sound pickup device 11 are connected to each other by a communication cable 12 and perform digital communication.
- the sound collector 11 transmits the audio signal collected by the microphone element 31 and the filter coefficient of the echo canceller 32 (see FIG. 3) to the communication device 10.
- the communication device 10 is connected to a personal computer 2 that is a host device.
- the personal computer 2 communicates with another voice conference system installed at another base via a network 3 such as the Internet, and is inputted from the communication device 10 of the voice conference system 1 (the microphone element 31 is collected).
- the sound signal is transmitted to another audio conference system, and the audio signal received from the other audio conference system is input to the communication device 10.
- the communication device 10 emits an audio signal sent from another audio conference system from the speaker 26.
- FIG. 2 is a diagram for explaining a form of sound reflection in the conference room C.
- FIG. The sound emitted from the speaker 26 directly reaches the conference participant M and the microphone element 31, and variously reflects on the wall and ceiling of the conference room C to reach the participant M and the microphone element 31.
- the so-called sound becomes a so-called echo that is played back.
- the microphone element 31 is connected to an echo canceller 32 (see FIG. 3) that cancels the sound emitted from the speaker 26.
- the communicator 10 includes a reverberation suppression filter. 24 (see FIG. 6) is provided.
- the filter coefficient of the filter 24 is calculated using the filter coefficient of the adaptive filter 35 (see FIG. 5) of the echo canceller 32.
- the function part incorporated in the communication device 10 and the sound collector 11 described below may be configured by an electronic circuit, or may be realized by cooperation of a processor such as a computer and a program.
- FIG. 3 is a block diagram of the sound collector 11.
- FIG. 4 is a diagram illustrating the directivity of each of the three microphone elements 31 of the sound collector 11.
- FIG. 5 is a block diagram of the echo canceller 32 of the sound collector 11.
- the sound collector 11 includes three microphone elements 31. As shown in FIGS. 1 and 4, the sound collector 11 has a disk-like planar shape, and three microphone elements 31 face outward (normal direction) at intervals of 120 degrees on the circumference. It is provided radially. Each microphone element 31 is a unidirectional microphone, and has a cardioid sound collection characteristic centering on the direction in which the microphone element 31 faces. The microphone elements 31 are provided at intervals of 120 degrees, and the directivity characteristics thereof are arranged as shown in FIG. 4. Therefore, if the collected sound signals of the microphone elements 31 are synthesized, a signal having characteristics almost omnidirectional is obtained. can get.
- the microphone element 31 is not limited to the one having directivity characteristics of cardioid. It may have some directivity behind it, or may be bi-directional.
- each microphone element 31 is provided with an echo canceller 32.
- the detailed configuration of the echo canceller 32 will be described with reference to FIG. 5, but cancels the sound emitted from the speaker 26 from the sound signal collected by the microphone element 31.
- An audio signal in which the wraparound sound from the speaker 26 is canceled by the echo canceller 32 is input to the audio selection unit 33.
- Audio signals picked up by the three microphone elements 31 are input to the audio selector 33.
- the voice selection unit 33 is estimated as the voice signal input from which microphone element 31 is high, that is, the voice signal of the speaker. And one speech signal estimated as the speech signal is selected.
- an optimal microphone element 31 is selected from the three microphone elements 31, and an uttered voice with a good S / N ratio is collected. is doing.
- the selected audio signal is transmitted to the communication device 10 via the communication interface 34.
- the communicator 10 converts the sound signal received from each sound collector 11 into the sound signal level and duration. Further, one voice signal is selected by comparing the correlation degrees, or a plurality of voice signals are mixed, and the selected voice signal or the mixed voice signal is transmitted to the partner system.
- FIG. 5 is a block diagram of the echo canceller 32.
- the echo canceller 32 includes an adaptive filter 35 having a filter coefficient setting unit 35B and a variable filter 35A, and further includes an adder 37.
- an adaptive filter is a filter that automatically adapts its own transfer function (adaptive filter coefficient sequence) in accordance with a predetermined optimization algorithm.
- the filter coefficient setting unit 35B estimates the transfer function of the acoustic transfer system (sound propagation path from the speaker 26 to the microphone element 31) of the conference room C, and sets a filter coefficient that becomes a filter of the estimated transfer function to the variable filter 35A. Set to.
- An audio signal (sound emission signal) emitted from the speaker 26 is input to the variable filter 35A. Since the transfer function of the variable filter 35A is a transfer function simulating the acoustic transfer system (sound propagation path from the speaker 26 to the microphone element 31) of the conference room C, the sound emission signal filtered by the variable filter 35A is This is an audio signal (pseudo-regression sound signal) that simulates an audio signal (regression sound signal) that is emitted from the speaker 26, propagates through the conference room C, and is collected by the microphone element 31. The pseudo regression sound signal is input to the adder 37.
- the adder 37 receives an audio signal (sound collection signal) picked up by the microphone element 31.
- the adder 37 subtracts the pseudo regression sound signal from the collected sound signal and outputs the result.
- the collected sound signal includes a return sound signal that is emitted from the speaker 26 and circulates along with the speech signal of the conference participant M.
- the adder 37 subtracts the pseudo-regression sound signal from the sound collection signal, thereby removing the return sound from the sound collection signal, that is, canceling the echo.
- the collected sound signal whose echo has been canceled is input to the sound selection unit 33 and also input to the filter coefficient setting unit 35B as a reference signal.
- the filter coefficient setting unit 35B also receives a sound emission signal that is a sound signal emitted from the speaker 26 as a reference signal.
- the filter coefficient setting unit 35B continuously updates the filter coefficient based on these reference signals.
- the filter coefficient is updated by automatically detecting a time interval in which the sound is emitted from the speaker 26 and the participant M in the conference room C is not speaking and referring to the time interval. This is done using signals.
- variable filter 35A is an FIR filter. Therefore, the filter coefficient set in the variable filter 35A is simulated by the filter coefficient setting unit 35B estimating the impulse response of the acoustic propagation path from the speaker 26 to the microphone element 31. The filter coefficient setting unit 35B transmits the filter coefficient as an estimated impulse response to the communication device 10 via the communication interface 34.
- the sound selection unit 33 selects one of the sound signals picked up by the three microphone elements 31 and transmitted to the communication device 10. All three estimated impulse responses are transmitted to the communicator 10. As will be described later, the parameter estimation unit 23 of the communication device 10 combines these three estimated impulse responses. As shown in FIG. 4, the three estimated impulse responses are impulse responses including a reverberation component that arrives from the direction in which the corresponding microphone element 31 faces, but by combining the three, they arrive from all directions. Impulse responses from all directions of the conference room C collected by an omnidirectional microphone including a reverberation component can be simulated.
- FIG. 6 is a block diagram of the communication device 10.
- the communication device 10 includes a communication interface 21 for communicating with the personal computer 2, a microphone mixer 22, a parameter estimation unit 23, a filter 24, an audio circuit 25, a speaker 26, and a communication interface 27 for communicating with the sound collector 11. .
- the communication interface 21 is an interface for performing digital communication with the personal computer 2.
- a USB interface is used.
- the personal computer 2 is a host and the communication device 10 is an audio device.
- a plurality of communication interfaces 27 are provided, and individual sound collectors 11 are connected to each other via the cables 12.
- a wired LAN interface may be used as the communication interface 27.
- the communication device 10 receives an audio signal (acquisition signal whose echo is canceled) and three estimated impulse responses from the sound collector 11 via the communication interface 27.
- the received audio signal is input to the microphone mixer 22.
- a plurality of audio signals received from different sound collectors 11 are input from the plurality of communication interfaces 27 to the microphone mixer 22.
- the microphone mixer 22 selects or mixes the audio signals received from the plurality of sound collectors 11 to form a monaural audio signal, and transmits the monaural audio signal to the personal computer 2 via the communication interface 21.
- the personal computer 2 transmits this audio signal to the audio conference system at another site via the network 3.
- the microphone mixer 22 may select a speech signal having a good S / N ratio as an uttered speech to be transmitted to the partner system by comparing the level, duration or correlation of the speech signal of the communication device.
- the personal computer 2 receives an audio signal from the audio conference system at another site.
- This audio signal is input via the communication interface 21, input to the filter 24 as a sound output signal output from the speaker 26, and transmitted to each sound collector 11 via the communication interface 27.
- the filter 24 performs a filter process that suppresses a decrease in the intelligibility of the voice due to the reverberation of the conference room C. That is, signal processing is performed on the sound emission signal so as to suppress the level of the frequency band having a long reverberation time. In particular, since reverberation in the low frequency range causes a decrease in clarity, the degree of suppression is increased for the low frequency range.
- Such filter coefficients are determined by the parameter estimation unit 23.
- the sound emission signal whose frequency band having a long reverberation time is suppressed by the filter 24 is input to the audio circuit 25.
- the audio circuit 25 converts the sound emission signal into an analog audio signal, amplifies it at a predetermined level, and inputs it to the speaker 26.
- the speaker 26 emits the sound emission signal as sound into the conference room C. The emitted sound is heard by the conference participant M and collected by the microphone element 31.
- the sound emission signal transmitted to the sound collector 11 via the communication interface 27 is input as a reference signal to the filter coefficient setting unit 35B of the echo canceller 32 shown in FIG.
- FIG. 7 is a block diagram of the parameter estimation unit 23.
- FIG. 8 is a diagram illustrating a procedure of dereverberation processing executed in the audio conference system 1 including the parameter estimation unit 23.
- FIGS. 9A and 9B are diagrams illustrating signal waveforms that appear in the reverberation suppression processing procedure.
- the sound collector 11 performs sound collection by the directional microphone element 31 (S101), echo cancellation processing (S102), and extraction of a filter coefficient (estimated impulse response) from the adaptive filter 35 (S103). Do.
- the sound collector 11 transmits the filter coefficients of three echo cancellers 32 provided corresponding to the three microphone elements 31 to the communication device 10 as estimated impulse responses.
- the parameter estimation unit 23 includes a filter coefficient integration unit 40 for each connected sound collector 11, a reverberation time estimation unit 41, a reverberation time integration unit 42 for each connected sound collector 11, and a correction.
- a characteristic calculation unit 43 and a filter coefficient calculation unit 44 are provided.
- the communication device 10 receives three estimated impulse responses (filter coefficients) from each sound collector 11.
- the received estimated impulse response is input to the parameter estimation unit 23.
- the input estimated impulse response is input to the filter coefficient integration unit 40 provided for each sound collector 11.
- the filter coefficient integration unit 40 synthesizes the three input estimated impulse responses with the time axis aligned. This synthesis may be simply addition synthesis, the weight of each estimated impulse response may be changed, or the time lag of each impulse response may be corrected.
- an impulse response including a reverberation component arriving from a wider range than the estimated impulse response for one microphone (ideally 360 degrees omnidirectional). Can be estimated.
- This process is the previous stage integration of S104 in FIG. This process is performed for each sound collector 11 connected (estimated impulse response is input), and the impulse response at the position of each sound collector 11 is estimated.
- the wide-directional estimated impulse response synthesized by the filter coefficient integration unit 40 is input to the reverberation time estimation unit 41.
- the reverberation time estimation unit 41 performs the following processing. First, the estimated impulse response is passed through a band-pass filter of a plurality of channels to divide the band. The number of channels to be divided and the frequency band of each channel are arbitrary. For example, band division such as dividing 315 Hz to 8000 Hz into 15 channels may be performed. By this processing, the impulse response of the signal component of each frequency band (channel) is estimated. This process is the process of S105 in FIG. This process is also performed for each sound collector 11.
- the reverberation time estimation unit 41 obtains the reverberation time of the signal in each frequency band based on the estimated impulse response in each frequency band.
- the reverberation time generally refers to the time until the signal level is attenuated to ⁇ 60 dB (parts per million), and there are various calculation / estimation methods.
- the reverberation time may be obtained by the Schrader method.
- a Schrader curve reverberation decay curve as shown in FIG. 9A is obtained by Schrader integration or backward cumulative addition of the impulse response, and the time until this curve reaches ⁇ 60 dB can be obtained. It ’s fine.
- a predetermined section that does not include the direct sound or error component of the Schrader curve may be taken out, and the slope of the section may be defined as the slope of this curve, and the time for attenuation from 0 dB to ⁇ 60 dB due to this slope may be estimated.
- This process corresponds to S106 in FIG. 8 and is performed for each frequency band for each sound collector 11, and the reverberation time for each frequency band at the position of each sound collector 11 is estimated.
- the reverberation time for each frequency band at the position of each sound collector 11 estimated by the plurality of reverberation time estimation units 41 is input to the reverberation time integration unit 42.
- the reverberation time integration unit 42 synthesizes the reverberation time at the position of each sound collector 11 for each frequency band. This process is the latter stage integration process of S107 in FIG. 8, and is performed for each frequency band.
- this post-integration process is performed by averaging the reverberation times of the sound collectors 11 for each frequency band, but reverberation time values (outliers) that are extremely far from the average value are excluded from the average. May be. Further, the sound collector 11 having many outliers may be installed in a place where a biased characteristic is likely to appear, such as a corner of a room, and the reverberation time of the sound collector 11 is the entire surface in all frequency bands. Specifically, it may be excluded from the subsequent integration process.
- This exclusion process may be performed based on the judgment of the communication device 10, and a person who installs this system in the conference room C manually operates the sound collection device 11 or the communication device 10 to specify a specific sound collection device 11, for example,
- the sound collectors 11 and the like installed in the corners of the room may be set so as to be excluded from the targets for subsequent integration. In this case, for the sound collector 11 excluded from the integration target, the processing of S103 and subsequent steps is not necessary, and the processing is reduced.
- the average reverberation characteristic of the entire conference room C as shown in FIG. 9B is obtained. Based on the reverberation characteristics, it is possible to determine in which frequency band the reverberation time is long.
- the reverberation characteristic obtained by the reverberation time integration unit 42 is input to the correction characteristic calculation unit 43. Based on the input reverberation characteristic, the correction characteristic calculation unit 43 suppresses a frequency band having a long reverberation time so that the sound emitted from the speaker 26 is not covered by the reverberation sound of the sound. Determine the correction characteristics.
- the correction characteristics are determined by setting a reverberation time threshold value for each frequency band, extracting a frequency band whose reverberation time exceeds the threshold value, and suppressing the power of this frequency band, or a known method for suppressing reverberation sound.
- a gain table for each frequency band By using a filtering method, a gain table for each frequency band, a method for determining a power suppression amount for each frequency band can be selected.
- a gain table for each frequency band as shown in FIG. 10 can be used.
- the vertical axis represents gain (dB) and the horizontal axis represents reverberation time RT (seconds), and the gain value for each frequency band is indicated by a line segment having a slope.
- the line segments f1 to fn correspond to the frequency bands divided by the bandpass filter described above, and f1 is on the low sound range side and fn is on the high sound range side. For example, if the reverberation time is 1.0 second in the band f3, the gain is determined to be ⁇ 30 dB.
- the low-frequency line segment is set to have a steeper slope.
- the gain is around -24 dB. In this way, when the reverberation time in the low sound region is long, a correction characteristic that suppresses the low sound region more strongly than in the case where the reverberation time in the high sound region is long is determined.
- the gain table may be set such that the convergence point where a plurality of line segments converge as shown in FIG. 11 is shifted in the positive direction by a certain reverberation time. In this case, when the reverberation time is 1.0 second or less, the gain is 0 dB. This process is S108 in FIG.
- the determined correction characteristic is input to the filter coefficient calculation unit 44.
- the filter coefficient calculation unit 44 determines the filter characteristic so that the filter 24 has the correction characteristic calculated by the correction characteristic calculation unit 43.
- the filter 24 is configured by an FIR filter or an IIR filter.
- the filter coefficient is calculated by an operation such as a discrete time inverse Fourier transform or a parametric peak filter in accordance with the configuration of the filter 24. This process is S109 of FIG.
- the calculated filter coefficient is set in the filter 24 (S110).
- the reverberation characteristics of the conference room C are estimated using the filter coefficient of the echo canceller 32, and the clarity of the emitted sound is prevented from being lowered by suppressing the frequency band having a long reverberation.
- the frequency characteristic of the conference room C is estimated using the filter coefficient of the echo canceller 32, and the frequency characteristic of the sound emission signal is changed to the frequency of the conference room C so that the emitted sound is heard with a flat characteristic. You may correct
- FIG. 12 shows a modification of the parameter estimation unit.
- the parameter estimation unit 23 ′ in this figure determines a correction characteristic for correcting the frequency characteristic in addition to the correction characteristic for suppressing reverberation and sets it in the filter 24.
- the parameter estimation unit 23 'in this figure further includes a frequency characteristic estimation unit 45 and a frequency characteristic integration unit 46 for each sound collector 11, in addition to the configuration of the parameter estimation unit 23 in FIG.
- the wide impulse (omnidirectional) estimated impulse response for each sound collector 11 output from the filter coefficient integration unit 40 is input to the reverberation time estimation unit 41 and also to the frequency characteristic estimation unit 45. .
- the frequency characteristic estimation unit 45 performs a Fourier transform on the input impulse response, and calculates the frequency characteristic at the position of the sound collector 11. This frequency characteristic is input to the frequency characteristic integration unit 46.
- the frequency characteristic integration unit 46 synthesizes the frequency characteristics in the sound collectors 11 input from the frequency characteristic estimation units 45 to calculate an average value of the entire frequency characteristics in the conference room C. The calculation of the average value may be simply an arithmetic average, or an average may be obtained after normalizing each frequency characteristic.
- the frequency characteristic of the conference room C obtained by the frequency characteristic integration unit 46 is input to the correction characteristic calculation unit 43 ′.
- the correction characteristic calculation unit 43 ′ is a characteristic that suppresses a frequency band having a long reverberation time, and cancels the frequency characteristic that the emitted sound is influenced by the conference room C and passes through a flat transfer characteristic. Calculate correction characteristics that reach the listener. In addition, this calculation method may calculate a correction characteristic that reaches the listener via a preset ideal transmission characteristic, instead of a correction characteristic that reaches the listener via a flat transfer characteristic.
- the calculated correction characteristic is input to the filter coefficient calculation unit 44.
- the filter coefficient calculation unit 44 determines the filter characteristic so that the filter 24 has the correction characteristic calculated by the correction characteristic calculation unit 43 ′.
- the calculated filter coefficient is set in the filter 24.
- the communication device 10 performs the previous stage integration, but the sound collector 11 may perform this integration.
- the processing of S101 to S103 is performed by the sound collector 11.
- the communication device 10 performs the processing after S107.
- the processing of S104 to S106 during that time may be performed by either the sound collector 11 or the communication device 10.
- the audio conference system in which the sound collector 11 including the microphone element 31 is connected to the communication device 10 including the speaker 26 has been described.
- the plurality of microphone elements 31 and the speaker 26 are integrated.
- the present invention can also be applied to the audio conference apparatus (only the communication device 10) provided in the above.
- connection form between the communication device 10 and the sound pickup device 11 is not limited to wired connection.
- a wireless connection such as a wireless LAN or a short-range wireless communication standard may be used.
- the shape of the sound collector 11 and the number of microphone elements 31 are not limited to those shown in FIGS.
- two or four microphone elements 31 may be provided at equal intervals on the peripheral edge of a disk-shaped housing.
- the angle of each microphone element 31 is 180 degrees or 90 degrees. Further, the interval (angle) may not be equal.
- the microphone element 31 may be provided so as to be shifted in the direction toward the conference participant M.
- a plurality of sound collectors 11 When a plurality of sound collectors 11 are connected to the communication device 10, a plurality of sound collectors 11 may be daisy chain connected with a cable 12 as shown in FIG. 13. By using this connection form, the total cable length can be saved.
- the communication interfaces 21 and 34 LAN interfaces it is possible to correspond to both the star type connection of FIG. 1 and the daisy chain type connection of FIG.
- a plurality of microphones 51, 52, 53, and 54 each including one microphone element 31 are combined (grouped), and one group 60 is assigned to one group 60.
- the sound collector 11 may function.
- the attendant may set the grouping information of the table microphone 51 in the communication device 10 in advance, and a signal distribution unit is provided in the front end of the communication device 10 so that the communication device 10 performs grouping by itself. You may do it.
- the signal distribution unit groups the table microphones collecting similar signals into the same group based on the time position of the adaptive filter of the echo canceller, the degree of correlation of the collected sound signals, and the like. You may do it.
- FIG. 14A shows an example in which a plurality of table microphones (stand microphones) 51 are combined to form a group 60.
- FIG. 14B shows an example in which a plurality of hand microphones 52 are grouped into 60 groups.
- the hand microphone 52 may be wired or wireless.
- a plurality of hand microphones 52 existing at a certain distance may be set as one group 60, and an adaptive filter coefficient sequence may be added for each group 60.
- the reverberation time described above may be obtained by calculating the reverberation time of each group.
- the presence of a plurality of hand microphones 52 at a certain distance is determined by detecting the position by calculating the difference in sound collection delay between the two hand microphones 52 and by detecting the intensity of radio waves emitted by the hand microphones 52. be able to.
- the sound collector 11 may not be placed on the conference table D. That is, as shown in FIG. 14C, a hanging microphone 53 suspended from the ceiling or a wall microphone 54 installed on the wall surface may be used.
- the sound collector 11, the table microphone 51, the hand microphone 52, the hanging microphone 53, and the wall surface microphone 54 of FIG. 1 may be mixed.
- a sound emission and collection device of the present invention includes a speaker, a filter that processes a sound emission signal that is an audio signal supplied to the speaker, a plurality of directional microphones, a plurality of echo cancellers, and a first integration unit.
- the plurality of echo cancellers are provided corresponding to each of the plurality of microphones, and each cancels a return sound signal of the sound emitted by the speaker from the sound collection signal of the corresponding microphone.
- the first integration unit integrates the adaptive filter coefficient sequences extracted from the echo canceller.
- the reverberation time estimation unit estimates the reverberation time for each frequency band in the space where the speaker and the microphone exist based on the integrated filter coefficient sequence.
- the calculation unit extracts a frequency band having a long reverberation time, calculates a filter coefficient that suppresses power in the frequency band, and sets the filter coefficient in the filter.
- the sound emission and collection device of this invention includes a plurality of microphones.
- the plurality of microphones are directional microphones suitable for, for example, a conference.
- Each microphone is provided with an echo canceller for canceling the echo of the speaker sound.
- the echo canceller includes an adaptive filter that generates a pseudo-regression sound signal, and has an adaptive filter coefficient (estimated impulse response) that simulates an impulse response between a speaker and a microphone. Further, the estimated impulse response is constantly updated based on the sound emission signal and the sound collection signal of the microphone.
- the microphone is a directional microphone
- this estimated impulse response contains only a large amount of reverberation components coming from the direction of the microphone's directivity, and cannot be said to completely represent the reverberation characteristics of the entire conference room.
- the first integration unit can simulate an impulse response including reverberation components coming from a wide range of directions, even though it is a directional microphone for conferences. .
- a reverberation time is calculated using the integrated parameter (estimated impulse response), and a filter coefficient that suppresses the reverberation is calculated.
- the reverberation characteristics of the entire conference room can be accurately reproduced, and effective reverberation can be suppressed.
- the estimated impulse response used by the echo canceller for removing the return sound can be used as it is, no special calculation amount is required, and for this reason, it is not necessary to emit the test sound.
- a plurality of microphones may be arranged in different directions so that any microphone has sensitivity in all horizontal directions.
- the calculation unit compares a reverberation time threshold set in advance for each frequency band with a reverberation time for each estimated frequency band, and the reverberation time exceeds the threshold. To extract the frequency band.
- the filter coefficient set by the arithmetic unit is stronger in power for suppressing the low frequency band in the spatial frequency band than in the power for suppressing the high frequency band in the spatial frequency band.
- the first integration unit integrates the adaptive filter coefficients by aligning their time axes and changing simple integration or weighting.
- a sound collector including a plurality of microphones and a plurality of echo cancellers, and a communication device including a speaker and a filter may be provided separately.
- the degree of freedom of installation is increased and a plurality of sound collectors can be provided.
- a plurality of sound collectors are provided, a plurality of first integration units are provided corresponding to the plurality of sound collectors, and a reverberation time estimation unit is further integrated with the reverberation time of each sound collector. May be further provided. Then, the calculation unit may calculate the filter coefficient based on the reverberation time integrated by the second integration unit.
- the at least one sound collector is a plurality of sound collectors
- a plurality of the first integration units are provided corresponding to the plurality of sound collectors
- the sound emitting and collecting apparatus includes the plurality of sound collectors.
- a plurality of frequency characteristic estimation units for calculating a plurality of frequency characteristics at positions of the plurality of sound collectors based on a plurality of adaptive filter coefficients respectively integrated by the first integration unit of the sounder; and the plurality of frequencies
- a frequency characteristic integration unit that integrates the plurality of frequency characteristics calculated by the characteristic estimation unit;
- the present invention it is possible to provide a sound emission and collection device that can suppress reverberation by using a sound collection and echo cancellation function for a conference without reproducing test audio in advance.
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Abstract
Description
本発明の放収音装置は、スピーカと、スピーカに供給される音声信号である放音信号を処理するフィルタと、指向性を有する複数のマイクと、複数のエコーキャンセラと、第1統合部と、残響特性推定部と、演算部と、を備える。複数のエコーキャンセラは、複数のマイクの各々に対応して設けられ、各々がその対応するマイクの収音信号からスピーカによって放音された音声の回帰音信号をキャンセルする。第1統合部は、エコーキャンセラから取り出された適応フィルタ係数列を統合する。残響時間推定部は、統合されたフィルタ係数列に基づき、スピーカおよびマイクが存在する空間の周波数帯域ごとの残響時間を推定する。演算部は、残響時間の長い周波数帯域を抽出し、その周波数帯域のパワーを抑制するフィルタ係数を算出してフィルタに設定する。
この発明の放収音装置は、複数のマイクを備える。前記複数のマイクは例えば会議などに好適化された指向性マイクである。各マイクには、スピーカ音のエコーをキャンセルするためのエコーキャンセラが設けられている。エコーキャンセラは、疑似回帰音信号を生成する適応フィルタを備え、スピーカとマイクとの間のインパルス応答を模擬した適応フィルタ係数(推定インパルス応答)を持つ。また、この推定インパルス応答は、放音信号およびマイクの収音信号に基づいて常に更新されている。マイクは指向性マイクであるため、この推定インパルス応答はマイクの指向性の方向から到来する残響成分しか多く含まず、会議室全体の残響特性を完全に現しているとはいえない。しかし、第1統合部が、複数の指向性マイクのパラメータを統合することにより、会議用の指向性マイクでありながら、広い範囲の方向から到来する残響成分を含むインパルス応答を模擬することができる。そして、この統合されたパラメータ(推定インパルス応答)を用いて残響時間を算出し、この残響を抑制するようなフィルタ係数を算出する。これにより、会議室全体の残響特性を正確に再現でき、効果的な残響の抑制ができる。また、推定インパルス応答は、エコーキャンセラが回帰音除去に使用するものがそのまま流用できるので、特別な計算量が必要にならず、また、このためにテスト音声を放音する必要も無い。
本発明は、2014年4月14日出願の日本特許出願(特願2014-083209)に基づくものであり、その内容はここに参照として取り込まれる。
Claims (10)
- スピーカと、
前記スピーカに供給される音声信号である放音信号を処理するフィルタと、
複数のマイクと、
前記複数のマイクの各々に対応して設けられ、各々がその対応するマイクの収音信号から前記スピーカによって放音された音声の回帰音信号をキャンセルする複数のエコーキャンセラと、
前記複数のエコーキャンセラから取り出された適応フィルタ係数を統合する第1統合部と、
統合された前記適応フィルタ係数に基づき、前記スピーカおよび前記複数のマイクが存在する空間の周波数帯域毎の残響時間を推定する残響時間推定部と、
前記推定された残響時間に基づいて、前記放音信号から残響時間の長い周波数帯域を特定し、該特定された周波数帯域のパワーを抑制するフィルタ係数を算出して前記フィルタに設定する演算部と、
を備えた放収音装置。 - 前記複数のマイクは、指向性を有する請求項1に記載の放収音装置。
- 前記複数のマイクは、水平の全方向にいずれかのマイクが感度を持つようそれぞれ異なる方向に向けて配置されている請求項1または2に記載の放収音装置。
- 前記演算部は、周波数帯域ごとに予め設定されている残響時間のしきい値と、前記推定された周波数帯域毎の残響時間とをそれぞれ比較し、前記残響時間が前記しきい値を超えている周波数帯域を抽出する請求項1から3のいずれか1項に記載の放収音装置。
- 前記演算部が設定するフィルタ係数は、前記空間の周波数帯域における高音域の周波数帯域を抑制するパワーよりも前記空間の周波数帯域における低音域の周波数帯域を抑制するパワーの方が強い請求項1から4のいずれか1項に記載の放収音装置。
- 前記第1統合部は、前記適応フィルタ係数をそれらの時間軸を揃えて、単純統合又は重み付けを変更して統合する請求項1から5のいずれか1項に記載の放収音装置。
- 前記複数のマイクおよび前記複数のエコーキャンセラが含まれる少なくとも一つの収音器と、前記スピーカおよび前記フィルタが含まれる少なくとも一つの通信器と、
を備え、
前記収音器及び前記通信器は別体に構成されている請求項1から6のいずれか1項に記載の放収音装置。 - 前記少なくとも一つの収音器は複数の収音器であり、
前記第1統合部は、前記複数の収音器に対応して複数設けられ、
前記残響時間推定部は、前記複数の収音器のそれぞれの残響時間を統合する第2統合部をさらに有し、
前記演算部は、前記第2統合部によって統合された残響時間に基づいて、前記フィルタ係数を算出する請求項7に記載の放収音装置。 - 前記少なくとも一つの収音器は複数の収音器であり、
前記第1統合部は、前記複数の収音器に対応して複数設けられ、
当該放収音装置は、
前記複数の収音器の前記第1統合部でそれぞれ統合された複数の適応フィルタ係数に基づいて、前記複数の収音器の位置における複数の周波数特性を算出する複数の周波数特性推定部と、
前記複数の周波数特性推定部で算出された前記複数の周波数特性を統合する周波数特性統合部と、
をさらに備える請求項7に記載の放収音装置。 - スピーカに供給される音声信号である放音信号をフィルタによって処理し、
複数のマイクの各々に対応して設けられた複数のエコーキャンセラによって、前記複数のマイクの収音信号から前記スピーカによって放音された音声の回帰音信号をキャンセルし、
前記複数のエコーキャンセラから取り出された適応フィルタ係数を統合し、
統合された前記適応フィルタ係数に基づき、前記スピーカおよび前記複数のマイクが存在する空間の周波数帯域毎の残響時間を推定し、
前記推定された残響時間に基づいて、前記放音信号から残響時間の長い周波数帯域を特定し、該特定された周波数帯域のパワーを抑制するフィルタ係数を算出して前記フィルタに設定する放収音方法。
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