JP4929685B2 - Remote conference equipment - Google Patents

Abstract

A teleconference device includes a loudspeaker array and microphone arrays arranged at the both sides of the loudspeaker array. A plurality of focal points are set in front of the respective microphone arrays and symmetrically with respect to the center line of the loudspeaker array. A flux of sound collection beams toward the focal points is outputted. By calculating a difference between the sound collection beams toward the focal points symmetric with respect to the center line, a sound component coming into the microphone from the loudspeaker array SPA is cancelled. Furthermore, a total of squares of wave height value of the difference value for a particular time is used to estimate which of the set focal points is the nearest. Furthermore, by comparing the totals of squares of the wave height values of the sound collection beams toward the focal points symmetric to each other, it is possible to judge the position of the speaker.

Description

  The present invention relates to an apparatus that includes a microphone array and a speaker array and reproduces received voice and its sound field, and more particularly to specifying the position of a speaker or a sound source from the microphone array.

  Conventionally, a means for receiving a sound on the transmission side and reproducing a sound field of the sound on the transmission side has been proposed (see Patent Documents 1 to 3). In such an apparatus, sound signals collected from a plurality of microphones or the like are transmitted, and a sound field on the transmission side is reproduced using a plurality of speakers on the reception side. This has the advantage that the position of the speaker can be specified by voice.

  Patent Document 1 discloses a method of creating stereoscopic audio information that reproduces a sound field of a transmission source by transmitting audio information received by a plurality of microphone arrays and outputting the same by the same number of speaker arrays. Yes.

  According to the method of Patent Document 1, it is possible to transmit the sound field itself of the transmission source without fail, and it is possible to specify the position of the speaker by voice, but there are problems such as using a lot of line resources. There are disclosed means for identifying and transmitting speaker position information (see, for example, Patent Document 2).

  In Patent Document 2, a speaker's voice is captured by a microphone, speaker position information is generated from speaker information obtained from the microphone, and the speaker position information is multiplexed and transmitted together with the voice information. An apparatus is disclosed that switches the position of a speaker to be ringed according to speaker position information sent on the receiving side and reproduces the voice and position of the speaker on the receiving side.

In Patent Document 3, since it is not practical to have each speaker have all microphones in a multi-person conference system, the phase of the audio signal input to each microphone is shifted using the microphone control unit. There is a description of a conference system that identifies speakers by synthesis. In Patent Document 3, the phase shift pattern corresponding to the seat position of the speaker is changed to determine the phase pattern that maximizes the voice, and the speaker position is specified from the determined phase shift pattern. Yes.
Japanese Patent Laid-Open No. 2-114799 JP-A-9-261351 JP-A-10-145663

  However, the above patent documents have the following problems.

  As described above, the method of Patent Document 1 has a problem of using a lot of line resources.

  In the methods of Patent Documents 2 and 3, it is possible to generate speaker position information based on speaker information obtained from a microphone, but this position detection is disturbed by the sound of a speaker that outputs the sound transmitted from the counterpart device. However, there is a problem that the microphone array (the camera in Patent Document 3) is pointed at by misinterpreting that the sound source is in a direction different from the actual direction.

  Accordingly, an object of the present invention is to enable a remote conference device to estimate a true sound source even if a speaker that outputs sound transmitted from a partner device wraps around a microphone and collects sound.

In the present invention, means for solving the above-described problems are configured as follows.

(1) The present invention includes a speaker array including a plurality of speakers for outputting audio, and the first and second microphone arrays provided so as to pick up the longitudinal sides of the speaker array, the first By synthesizing the audio signals picked up by the microphones of the microphone array and the second microphone array by delay processing, a plurality of first condensers are symmetric with respect to the longitudinal center line of the speaker array. A sound collection area setting means for setting a sound area and a plurality of second sound collection areas, and a sound signal collected from each of the plurality of first sound collection areas and the plurality of second sound collection areas; first sound source for selecting a difference signal calculating means for calculating respective difference signal of the audio signal picked up from the serial symmetric positions sound-pickup area pairs, the signal strength is greater sound-pickup area pair of the differential signal The sound collection area with the higher intensity of the collected sound signal is selected from the pair of sound collection areas selected by the position estimation means and the first sound source position estimation means, and the sound collection position is in this sound collection area. Second sound source position estimating means for estimating the sound collection area, and the sound collection area setting means further sets a plurality of narrow sound collection areas in the sound collection area selected by the second sound source position estimation means. And a function of generating a plurality of narrow sound collection beams each focused on the plurality of narrow sound collection areas, and a sound source in an area where the intensity of the collected sound signal is large among the plurality of narrow sound collection areas Third sound source position estimating means for estimating that there is a position is provided .

The sound collection area setting means sets a symmetrical position as the sound collection area, and generates the first and second sound collection beams by focusing on the sound collection area. Further, the sound transmitted from the counterpart device and output from the speaker array is output substantially symmetrically on either side of each of the pair of microphone arrays. Therefore, it is considered that the sound output from the speaker array is input to the first and second sound collecting beams substantially equally, and the difference signal calculating means calculates the difference signal of the first and second sound collecting beams. Therefore, the sound output from the speaker array can be canceled. In addition, even if the difference between the effective values of the collected sound beam is calculated, it is considered that the sound output from the speaker array is input almost equally to the focal point to which the collected sound beam is applied. Canceled audio can be canceled.

  Further, even if such a difference is taken, the sound other than the sound output from the speaker array input to the microphone array does not disappear. For example, typically, when a speaker speaks only to one microphone array side and a sound collecting beam directed toward the speaker is generated, one sound collecting beam includes Since the voice of the speaker enters and no voice is input to the opposite side, the voice of the speaker itself or the voice of the opposite phase remains in the calculation of the difference. Even if there are sound sources on both sides, the sound is different, and in most cases, the sound input to the pair of microphone arrays is asymmetric. Therefore, even if such a difference is taken, the speaker's voice remains. Further, even if the effective value is calculated, the presence of the speaker's voice can be similarly extracted.

  The first sound source position estimating means estimates that a sound source position exists in one of the sound collection area pairs having a large difference signal. The second sound source position estimating means compares the sound signals picked up in each of the sound pickup area pairs and estimates which sound source position exists. As described above, according to the present invention, the position of the sound source (including the voice of the speaker; the same shall apply hereinafter) is correctly estimated even when the sound output from the speaker may be collected around the microphone. can do.

  The effective value of the audio signal can be obtained by calculating in real time the time average of the square of the peak value at a specific time. The signal intensity of the difference signal is compared with the time average of the square of the peak value of a predetermined time, the square sum of a plurality of predetermined frequency gains of the FFT-transformed gain, or the like. The signal strength of the effective value difference signal can be calculated by using the time average of the effective value difference signal or the squared time average of the difference signal using data for a predetermined time longer than the effective value calculation. it can. The same applies hereinafter.

  In the present invention, a plurality of narrow sound collection areas are set in the sound collection area estimated by the second sound source position estimation means to have a sound source position, and a narrow sound collection beam is generated in each of the narrow sound collection areas. The third sound source position estimating means selects the area having a high signal intensity from the narrow sound collection areas, thereby narrowing down the position of the sound source in stages and estimating the position of the sound source in a short time rather than starting from the beginning. Can be estimated.

  According to the present invention, it is possible to correctly estimate the position of a sound source even when there is a possibility that the sound output from the speaker in the remote conference device may be collected by the microphone.

<First Embodiment>
The configuration and usage of the remote conference apparatus according to the first embodiment of the present invention will be described with reference to FIG. The remote conference apparatus according to the first embodiment reproduces and outputs the voice transmitted from the partner apparatus by reproducing the position of the speaker on the partner apparatus side using the speaker array, and uses the microphone array. , And the position of the speaker is detected, and the collected sound and position information are transmitted to the partner device.

FIG. 1 shows an external view and a usage form of the remote conference device. FIG. 1A is an external perspective view of the remote conference device, and FIG. 1B is a bottom view AA arrow of the remote conference device. FIG. FIG. 1C is a diagram showing a usage pattern of the remote conference apparatus.

  As shown in FIG. 1A, the remote conference apparatus 1 includes an apparatus body having a rectangular parallelepiped shape and legs 11. The main body of the remote conference apparatus 1 is supported by the legs 11 so as to float above the installation surface by a predetermined distance. On the bottom surface of the teleconferencing device 1, a speaker array SPA in which a plurality of speakers SP1 to SP4 are linearly arranged in the longitudinal direction of the device body which is a rectangular parallelepiped is provided downward. With this speaker array SPA, sound is output downward from the bottom surface of the remote conference apparatus 1, and this sound is reflected on the installation surface of the conference desk or the like and reaches the conference participants (see FIG. 1C).

  Further, as shown in FIGS. 1A and 1B, both side surfaces of the apparatus main body in the longitudinal direction (hereinafter, both side surfaces are referred to as right side surface (upper side of FIG. 1B) and left side surface (FIG. 1B). In the lower side), a microphone array in which microphones are arranged in a straight line is provided. That is, a microphone array MR including microphones MR1 to MR4 is provided on the right side surface of the apparatus main body, and a microphone array ML including microphones ML1 to ML4 is provided on the left side surface of the apparatus main body. Using the microphone arrays MR and ML, the remote conference apparatus 1 collects the voice of the conference participant who is a speaker and detects the position of the speaker.

  Although not shown in FIG. 1, inside the teleconference device 1, the voice collected from the microphone arrays MR and ML is processed, and the position of the speaker (not only the human voice but also the object) The same applies to the following, and the transmission unit 2 (see FIG. 4) for multiplexing and transmitting this position and the sound collected from the microphone arrays MR and ML, and receiving from the counterpart device The receiving unit 3 (see FIG. 6) is provided that converts the sound thus generated from the speakers SP1 to SP4 into a beam.

  In FIG. 1, the microphone arrays MR and ML are provided at symmetrical positions with respect to the center line 101 of the speaker array SPA. However, the apparatus according to the first embodiment does not necessarily have to be provided symmetrically. Even if the microphone arrays MR and ML are left-right asymmetric, signal processing may be performed by the transmission unit (see FIG. 4) so that the left and right sound collection areas (see FIG. 3) are formed symmetrically. .

  Next, a usage pattern of the remote conference apparatus 1 will be described with reference to FIG. The remote conference apparatus 1 is usually used in the center of the conference desk 100. Speakers 998 and / or speakers 999 are seated on the left and right sides or one side of the conference desk 100. The sound output by the speaker array SPA is reflected by the conference desk 100 and reaches the left and right speakers. The speaker array SPA converts the sound into a beam and outputs it, so that the right and left speakers are identified. Can be localized in position. Details of the sound beam conversion processing by the speaker array SPA will be described later.

  Further, the microphone arrays MR and ML pick up the voice of the speaker. A signal processing unit (transmission unit) connected to the microphone arrays MR and ML detects the position of the speaker based on the timing difference of the voices input to the microphone units MR1 to MR4 and ML1 to ML4.

  In FIG. 1, for ease of illustration, the number of speakers and the number of microphones are four. However, in order to use the apparatus of the first embodiment, the number of speakers is not limited to four, and one or many speakers are used. Microphones may be provided, and the microphone arrays MR, ML, and speaker array SPA may be provided in a plurality of rows instead of one row. Therefore, in the following description, for example, a speaker array and a microphone array using the suffix i in the manner of SPi (i = 1 to N) for the speakers SP1 to SPN and MLi (i = 1 to N) for the microphones ML1 to MLN. Each speaker and microphone will be expressed. For example, SPi (i = 1 to N) and i = 1 corresponds to SP1.

  Here, with reference to FIG. 2, the sound beam conversion processing by the speaker array SPA, that is, the sound beam and the sound collecting beam formed by the microphone arrays ML and MR will be described.

  FIG. 3A is a diagram for explaining an audio beam. The signal processing unit (receiving unit) that supplies audio signals to the speaker units SP1 to SPN of the speaker array SPA delays the audio signals received from the counterpart device by delay times DS1 to DSN as shown in FIG. It supplies to each speaker unit SP1-SPN. In this figure, the speakers closest to the virtual sound source position (focal point FS) emit sound without a delay time, and the sound is emitted through a delay time corresponding to the distance as the distance from the virtual sound source position increases. Such a delay pattern is given. Due to this delay pattern, the sound output from each of the speaker units SP1 to SPN spreads to form a wavefront similar to the sound emitted from the virtual sound source of FIG. The voice can be heard as if the other speaker is at the position of the virtual sound source.

  FIG. 5B is a diagram for explaining the sound collection beam. The audio signals input to the microphone units MR1 to MRN are synthesized after being delayed by delay times DM1 to DMN, respectively, as shown. In this figure, the sound signal picked up by each microphone is input to the adder without delay and the sound picked up by the microphone farthest from the sound pickup area (focal point FM) is closer to the sound pickup area. A delay pattern is provided so as to be input to the adder after being delayed for a time corresponding to the distance. Due to this delay pattern, each sound signal is equidistant from the sound collection area (focal point FM) in sound wave propagation, and each synthesized sound signal emphasizes the sound signal in this sound collection area with the same phase, The audio signal in the area is offset by the phase shift. In this way, by synthesizing the sound input to a plurality of microphones by delaying them so as to be equidistant in sound wave propagation from a certain sound collection area, only the sound in the sound collection area can be collected.

  In the remote conference apparatus of the present embodiment, each microphone array MR, ML forms a sound collecting beam simultaneously with respect to a plurality (four in FIG. 3) of sound collecting areas. As a result, the voice can be picked up wherever the speaker is in the sound pickup area, and the position of the speaker can be detected from the sound pickup area where the sound is picked up.

  Next, the detection of the sound source position by the sound collection beam and the sound collection operation from the sound source position will be described with reference to FIG. FIG. 3 is a plan view of the teleconference device and the speaker looking down from above, that is, a view taken along the line BB in FIG.

≪Description of sound source position detection and sound collection method excluding daemon sound source≫
First, the principle of the sound source position detection and sound collection method of the remote conference apparatus will be described. In this description, it is assumed that no sound beam is output from the speaker array SPA.

  Here, the process for the collected sound signal of the microphone array MR on the right side will be described. The transmission unit 2 (see FIG. 4) of the teleconference apparatus 1 forms a sound collection beam that focuses on the four areas of the sound collection areas 411 to 414 by the delay synthesis described above. The plurality of sound collection areas are determined on the assumption of a position where a speaker attending a conference using the remote conference device 1 may exist.

  Of these sound collection areas 411R to 414R, it is considered that a speaker (sound source) exists in an area where the level of the collected sound signal is the highest. For example, as shown in FIG. 3, when the sound source 999 is present in the sound collection area 414R, the sound signal collected from the sound collection area 414R compared to the sound signals collected from the other sound collection areas 411R to 413R. The level of will increase.

  Similarly, for the microphone array ML on the left side surface, four sound collecting beams are formed almost symmetrically with the right side surface, and the area where the level of the collected sound signal is the highest among the sound collecting areas 411L to 414L. To detect. The line symmetry line is formed so as to substantially coincide with the axis of the speaker array SPA.

  The above is the principle of the sound source position detection and sound collection method of the remote conference apparatus of this embodiment.

  In the state where no sound is output from the speaker array SPA and the microphone arrays MR and ML do not collect the sneak sound, the sound source position can be detected and collected in accordance with this principle. Audio signals are transmitted and received, and sound is emitted from the speaker array SPA in parallel with sound collection by the microphone arrays MR and ML.

  The sound signal supplied to each speaker of the speaker array SPA has a pattern delay as shown in FIG. 2A so as to form the same wavefront as when sound comes from the virtual sound source position set behind the speaker array. Is given. On the other hand, the audio signal collected by the microphone array MR is synthesized after being delayed by a pattern as shown in FIG. 2B so that the timing of the audio signal coming from a predetermined sound collection area matches.

  Here, when the virtual sound source position of the speaker array SPA coincides with any one of the plurality of sound collection areas of the microphone array MR, the delay pattern given to each speaker SP1 to SPN of the speaker array SPA and the microphone array MR. The delay pattern assigned to the sound collection area for the sound signal picked up by the sound is just reversed, and the sound signal that is emitted from the speaker array SPA and wraps around the microphone array MR is synthesized at a high level. Will be.

  In the case of processing by the general sound source position detection method described above, there is a problem that the wraparound audio signal synthesized at this large level is erroneously recognized as an original sound source (daemon sound source).

  Therefore, if this demon sound source is not canceled, the voice signal arriving from the counterpart device is returned as it is, causing echoes, and the sound of the original sound source (speaker) cannot be detected and collected. .

  The above is the description of the microphone array MR, but the same applies to the microphone array ML (because it is symmetrical).

  That is, the demon sound source is generated symmetrically in the same manner in the right microphone array MR and the left microphone array ML because the sound beam is reflected by the conference desk 100 and radiated left and right symmetrically.

  Therefore, the sound volume levels of the left sound collecting areas 411L to 414L and the right sound collecting areas 411R to 414R are compared, and even if it is estimated that the sound volume exists and the sound volume is present, the sound volume levels are similarly large in the corresponding areas on the left and right. In this case, it is assumed that this is a demon sound source in which the sound beam of the speaker array SPA is circulated, and by removing this from the sound collection target, it is possible to detect and collect sound from the true sound source and Echoes due to voice are prevented.

  Therefore, in the transmission unit of this teleconference device, the sound signal level collected from the sound collection areas 411L to 414L of the left microphone array ML and the sound signal level collected from the sound collection areas 411R to 414R of the right microphone array MR If the levels of the left and right sound collection areas are significantly different, it is determined that a sound source exists in the larger sound collection area. Yes.

  Then, only the larger audio signal is transmitted to the counterpart device, and position information indicating the position of the sound collection area where the audio signal is detected is added to the subcode of the signal (digital signal).

  Hereinafter, the configuration of the signal processing unit (transmission unit) that executes the daemon sound source exclusion process will be described. The narrow sound collecting beams 431 to 434 in FIG. 3 will be described with reference to the description of the second embodiment in FIG.

≪Configuration of transmitter that forms sound collection beam≫
FIG. 4 is a block diagram illustrating a configuration of the transmission unit 2 of the remote conference device 1. Here, a thick arrow indicates that a plurality of audio signals are transmitted, and a thin arrow indicates that one audio signal is transmitted. A broken arrow indicates that an instruction input is transmitted.

  The first beam generation unit 231 and the second beam generation unit 232 in the figure are signals that form four systems of sound collection beams that focus on the left and right sound collection areas 411R to 414R and 411L to 414L shown in FIG. It is a processing unit.

  The first beam generator 231 receives an audio signal picked up by each microphone unit MR1 to MRN of the right microphone array MR via the A / D converter 211. Similarly, the second beam generation unit 232 receives an audio signal collected by the microphone units ML1 to MLN of the left microphone array ML via the A / D converter 212.

  The first beam generation unit 231 and the second beam generation unit 232 form four sound collection beams to collect sound from the four sound collection areas 411R to 414R and 411L to 414L, and the collected sound signals Is output to the difference value calculation circuit 22 and the selectors 271 and 272.

  FIG. 5 is a diagram illustrating a detailed configuration of the first beam forming unit 231. The first beam generation unit 231 includes a plurality of delay processing units 45j corresponding to the sound collection areas 41j (j = 1 to K). Each delay processing unit 45j delays an audio signal for each microphone output based on the delay pattern data 40j in order to generate a sound collection beam output MBj having a focus on each sound collection area 41j. Each delay processing unit 45j inputs the delay pattern data 40j stored in the ROM, and sets the delay amount in the delay 46ji (j = 1 to K, i = 1 to N).

  Then, the adder 47j adds these delayed digital audio signals and outputs the result as a microphone beam output MBj (j = 1 to K). Each of the sound collection beam outputs MBj is a sound collection beam that focuses on the sound collection area 41j shown in FIG. The sound collection beam output MBj calculated by each delay processing unit 45j is output to the difference value calculation circuit 22 and the like.

  In addition, although the first beam forming unit 231 has been described with reference to FIG. 5, the second beam forming unit 232 has the same configuration.

In FIG. 4, the difference value calculation circuit 22 compares the sound volume levels of the sound signals collected in the sound collection areas at the left and right symmetrical positions among the sound signals collected in each sound collection area, and calculates the difference value. To do. That is, if the signal level of the sound collection area A is represented by P (A), the difference value calculation circuit 22
D (411) = | P (411R) −P (411L) |
D (412) = | P (412R) −P (412L) |
D (413) = | P (413R) −P (413L) |
D (414) = | P (414R) −P (414L) |
Calculate The calculated difference values D (411) to D (414) are output to the first estimation unit 251.

  The difference value calculation circuit 22 may be configured to output the difference value signal by directly subtracting the signal waveform of the sound signal collected in the left and right sound collection areas. A value obtained by subtracting the volume level value obtained by integrating the effective value of the audio signal for a predetermined time may be output at each predetermined time.

  When the difference value calculation circuit 22 outputs a difference value signal, a BPF 241 may be inserted between the difference value calculation circuit 22 and the first estimation unit 251 in order to facilitate estimation by the first estimation unit 251. . The BPF 241 is set so as to pass the frequency band around 1 kHz to 2 kHz where the directivity control can be satisfactorily performed by the sound collection beam in the frequency range of the conversational sound from the difference value signal.

  As described above, the left and right microphone arrays MR and ML are differentiated from the speaker array SPA by subtracting the volume levels of the collected sound signals in the left and right sound collection areas at symmetrical positions with the center line of the speaker array SPA as the symmetry axis. The sound component that circulates symmetrically to the left and right is canceled, and the circulated sound signal is not erroneously recognized as a daemon sound source.

  The first estimation unit 251 selects the maximum difference value input from the difference value calculation circuit 22 and selects a pair of sound collection areas in which the maximum difference value is calculated. In order to input this sound collection area to the second estimation unit 252, the first estimation unit 251 outputs a selection signal for outputting the sound signal of this sound collection area to the second estimation unit 252 to the selectors 271 and 272.

  Based on this selection signal, the selector 271 selects the signal of the sound collection area selected by the first estimation unit 251 from the four sound collection area signals collected by the right beam generation unit 231 as the second estimation unit 252. A signal is selected to be supplied to the signal selector 26. Further, the selector 272 performs second estimation on the sound collection area signal selected by the first estimation unit 251 among the four sound collection area signals collected by the left beam generation unit 232 as a beam based on the selection signal. A signal is selected to be supplied to the unit 252 and the signal selection unit 26.

  The second estimation unit 252 receives the sound signal of the sound collection area estimated by the first estimation unit 251 and selectively output from the selectors 271 and 272. The second estimation unit 252 compares the input audio signals of the left and right sound collection areas, and determines that the higher level is the audio signal of the true sound source. The second estimation unit 252 outputs information indicating the direction and distance of the sound collection area where the true sound source is present to the multiplexing unit 28 as position information 2522, and also outputs the audio signal of the true sound source to the signal selection unit 26. Is selectively input to the multiplexer 28.

  The multiplexing unit 28 multiplexes the position information 2522 input from the second estimation unit 252 and the sound signal 261 of the true sound source selected from the signal selection unit 26, and the multiplexed signal is transmitted to the counterpart device. To send.

  Note that these estimation units 251 and 252 repeatedly perform estimation of the sound source position at regular intervals. For example, repeat every 0.5 seconds. In this case, the signal waveform or the effective amplitude value for 0.5 seconds may be compared. In this way, if the sound collection area is switched by repeatedly estimating the sound source position every predetermined period, it is possible to collect sound corresponding to the movement of the speaker.

  When the true sound source position and the demon sound source position due to wraparound overlap, a difference signal obtained by subtracting the left and right signal waveforms may be output to the counterpart device as a sound collection signal. This is because the difference signal cancels only the daemon sound source waveform and stores the signal waveform of the true sound source.

  Further, in order to cope with the case where the speaker exists over two sound collection areas or the case where the speaker moves, the following other forms are also conceivable. The first estimation unit 251 selects two sound collection areas in descending order of the intensity of the difference signal, and outputs the intensity ratio. The second estimation unit 252 compares the maximum pair or two pairs of signal strengths to estimate which side the true sound source is on. The signal selection unit 26 synthesizes the two audio signals on one side selected by the first estimation unit 251 and the second estimation unit 252 by applying the weight of the instructed intensity ratio, and outputs the resultant as an output signal 261 To do. In this way, if the voices at two positions are always synthesized with the weight of the signal intensity ratio, the same crossfade as described above is always applied to the movement of the speaker, and the sound image localization moves naturally.

<< Configuration of the receiving unit 3 that forms an audio beam >>
Next, the internal configuration of the receiving unit 3 will be described with reference to FIG. The receiving unit 3 receives the audio signal from the counterpart device, and localizes the audio signal from the audio signal receiving unit 31 that separates the position information from the subcode of the audio signal, and the position information separated by the audio signal receiving unit 31. A parameter calculation unit 32 that calculates a directivity control parameter for determining the position and localizing the sound image at the position, and controls the directivity of the received audio signal based on the parameter input from the parameter calculation unit 32. From the directivity control unit 33, the D / A converter 34i (i = 1 to N) for converting the sound signal whose directivity is controlled to the analog signal, and the D / A converter 34i (i = 1 to N). And an amplifier 35i (i = 1 to N) for amplifying the output analog audio signal. The analog audio signal output from the amplifier 35i is supplied to the external speaker SPi (i = 1 to N) shown in FIG.

  The audio signal receiving unit 31 is a functional unit that communicates with a partner apparatus via the Internet, a public telephone line, or the like, and includes a communication interface, a buffer memory, and the like. The audio signal receiving unit 31 receives the audio signal 30 including the position information 2522 as a subcode from the counterpart device. The position information is separated from the subcode of the received audio signal and input to the parameter calculation unit 32, and the audio signal is input to the directivity control unit 33.

  The parameter calculation unit 32 is a calculation unit that calculates parameters used in the directivity control unit 33. The parameter calculation unit 32 generates a focus at a position based on the received position information, and a sound signal is emitted from the focus. The amount of delay to be given to the audio signal supplied to each speaker unit is calculated in order to provide such directivity.

  The directivity control unit 33 processes the audio signal received by the audio signal reception unit 31 for each output system of the speakers SPi (i = 1 to N) based on the parameters set by the parameter calculation unit 32. That is, a plurality of processing units corresponding to each of the speakers SPi (i = 1 to N) are provided in parallel. Each processing unit sets a delay amount or the like for the audio signal on the basis of the parameter (delay amount parameter or the like) calculated by the parameter calculation unit 32, and each D / A converter 34i (i = 1 to N) is set. Output.

  The D / A converter 34 i (i = 1 to N) converts the digital audio signal for each output system output from the directivity control unit 33 into an analog signal and outputs the analog signal. The amplifier 35i (i = 1 to N) amplifies the analog audio signal output from the D / A converter 34i (i = 1 to N), and outputs the amplified signal to the speaker SPi (i = 1 to N). .

  In order for the receiving unit 3 described above to reproduce the positional relationship of the sound source in the partner device with the sound signal received from the partner device, the voice signal is received from the speaker array SPA installed on the bottom surface of the device body. Based on the above, a beam is generated and output, and the directivity as if sound is output from a virtual sound source position is reproduced.

<Second Embodiment>
Next, a remote conference apparatus according to the second embodiment will be described with reference to FIG. This embodiment is an application of the first embodiment shown in FIG. 4, and the same portions are denoted by the same reference numerals, and the description will be applied mutatis mutandis. Further, FIG. 3 is referred to supplementarily in the description of the sound collecting beam.

  In the first embodiment, it is assumed that a true sound source exists in one of the pairs of sound collection areas where the difference signal is large, and the second estimation unit 252 estimates in which one the true sound source exists. In the embodiment, further, a detailed position search beam (narrow beam) generation for accurately detecting the sound source position by searching in more detail the sound collection area where the true sound source estimated by the second estimation unit 252 exists. Functions 2313 and 2323 are provided.

  When the second estimation unit 252 estimates that the true sound source 999 is present in the sound collection area 414R as illustrated in FIG. 3, the second estimation unit 252 notifies the first beam generation unit 231 of the estimation result. . As described above, since the second estimation unit 252 estimates which side of the microphone array MR or ML has the true sound source, the notifications 2523 and 2524 of the estimation results are input to only one of them. If it is estimated that a true sound source is present in the left area, the second estimation unit 252 notifies the second beam generation unit 232 of the estimation result.

  Based on this notification, the first beam generation unit 231 operates the detailed position search beam generation function 2313 to generate a narrow beam focusing on the narrow sound collection areas 431 to 434 in FIG. The position of the sound source 999 is searched.

  Further, the apparatus according to the second embodiment includes a third estimation unit 253 and a fourth estimation unit 254. Two of the collected sound beams of the detailed position search beam generation functions 2313 and 2323 are selected in descending order of signal intensity. However, only the side estimated by the second estimation unit 252 operates among the estimation units 253 and 254.

  In the example of FIG. 3, the sound signal is collected from the sound collection beam directed to the narrow sound collection areas 431 to 434, and the true sound source 999 straddles the sound collection area 434 and the sound collection area 433. Exists in position. In this case, the third estimation unit 253 selects the sound signals collected from the sound collection areas 434 and 433 in descending order of signal strength. The third estimation unit 253 estimates and outputs the position of the speaker by proportionally allocating the focal position of the selected sound collection area according to the signal strengths of the two selected sound signals, and the two selected sound signals. The signals are weighted and synthesized and output as an audio signal.

  The first beam generation unit 231 (detail position search beam generation function 2313) and the third estimation unit 253 in the right area have been described above, but the second beam formation unit 232 (detail position search beam generation function in the left area) has been described. 2323) and the fourth estimation unit 254 have the same configuration and execute similar processing operations.

  Note that the detailed position search function of the apparatus of the second embodiment described above may not be able to catch up when the speaker moves frequently. Therefore, it is also conceivable to use this function only when the position of the speaker output from the second estimation unit 252 remains for a certain period of time. In this case, when the position of the speaker output from the second estimation unit 252 moves within a certain time, even if the configuration shown in FIG. 7 is provided, it is the same as in the first embodiment shown in FIG. It is sufficient to perform the operation.

  Note that the estimation units 253 and 254 that perform the narrowing estimation correspond to the “third sound source position estimation unit” of the present invention.

<Third Embodiment>
Next, the transmission unit of the remote conference apparatus according to the third embodiment of the present invention will be described with reference to FIG. FIG. 8 is a block diagram of this transmission unit. The transmission unit 2 of the apparatus of this embodiment generates a sound collection beam by using the output of the difference value calculation circuit 22 in that the input of the difference value calculation circuit 22 is the output of the A / D converters 211 and 212. The difference is that the third beam generation unit 237 is provided, the fourth beam generation unit 238 and the fifth beam generation unit 239 are provided, and the selectors 271 and 272 are not provided. The other parts are denoted by the same reference numerals, and the above description is applied mutatis mutandis. Only the differences and important points of the apparatus of this embodiment will be described below.

  As shown in FIG. 8, the outputs of the A / D converters 211 and 212 are directly input to the difference value calculation circuit 22. Therefore, in the apparatus according to the second embodiment, the number N of the microphones MRi and the microphones MLi are the same and are provided at symmetrical positions. The difference value calculation circuit 22 calculates “(voice signal of microphone MRi) − (voice signal of microphone MLi)” (i = 1 to N), respectively. As a result, as with the apparatus of the embodiment shown in FIG. 4, it is possible to cancel the amount of audio that has entered from the speaker array SPA and input to the microphone arrays MR and ML.

  Here, in the apparatus of the third embodiment, each of the microphones MRi and MLi needs to be substantially symmetrical with respect to the center line in the longitudinal direction of the speaker array SPA. This is because the difference value calculation circuit 22 cancels the wraparound sound between the microphones. The difference value calculation circuit 22 always performs calculation while the microphone arrays MR and ML of the remote conference apparatus 1 are activated.

  The third beam generation unit 237 focuses four virtual sound collection areas based on the bundle of output signals of the difference value calculation circuit 22 in the same manner as the first beam generation unit 231 and the second beam generation unit 232. The sound collection beam is output. This virtual sound collection area corresponds to a pair of sound collection areas (411R and 411L, 412R and 412L, 413R and 413L, 414R and 414L: see FIG. 3) set symmetrically with respect to the center line 101 of the speaker array SPA. The audio signal output from the third beam generation unit 237 is the same as the differential signals D (411), D (412), D (413), and D (414) in the first embodiment. If this difference signal is output to the first estimation unit 251 through the BPF 241, the sound source position can be estimated in the same manner as the first estimation unit 251 of the apparatus shown in FIG. The estimation results 2511 and 2512 are output to the fourth beam generation unit 238 and the fifth beam generation unit 239.

  The fourth beam generation unit 238 and the fifth beam generation unit 239 in FIG. 8 will be described. Digital audio signals output from the A / D converters 211 and 212 are directly input to the fourth beam generation unit 238 and the fifth beam generation unit 239. Based on this digital audio signal, a sound collection beam with the sound collection area indicated by the estimation results 2511 and 2512 input from the first estimation unit 251 as a focus is generated, and the sound signal in the sound collection area is extracted. That is, the sound collection beams generated by the fourth beam generation unit 238 and the fifth beam generation unit 239 correspond to the sound collection beams selected by the selectors 271 and 272 in the first embodiment.

  As described above, the fourth beam generation unit 238 and the fifth beam generation unit 239 output only one system of sound output collected by the sound collection beam instructed by the first estimation unit 251. The sound signals collected by the fourth beam generation unit 238 and the fifth beam generation unit 239 from the sound collection area that is the focal point of each sound collection beam are input to the second estimation unit 252.

  The following operations are the same as those in the first embodiment. The second estimation unit 252 compares the two audio signals and determines that a sound source exists in the sound collection area having the higher level. The second estimation unit 252 outputs information indicating the direction and distance of the sound collection area where the true sound source is present to the multiplexing unit 28 as position information 2522, and also outputs the audio signal of the true sound source to the signal selection unit 26. Is selectively input to the multiplexer 28. The multiplexing unit 28 multiplexes the position information 2522 input from the second estimation unit 252 and the sound signal 261 of the true sound source selected from the signal selection unit 26, and the multiplexed signal is transmitted to the counterpart device. To send.

  In the third embodiment shown in FIG. 8 as well, as in the second embodiment, it is possible to perform a multi-stage estimation and search by narrowing down the position of the sound source at first wide and narrow again. . In that case, when one search is completed, the second estimation unit 252 outputs instruction inputs 2523 and 2524 for instructing to search a narrower range to the fourth and fifth beam generation units 238 and 239. This operation is output only to the beam generator on the side where the sound source exists. Upon receiving this instruction input, the beam generation unit reads out the delay pattern corresponding to a narrower internal range and rewrites the delay pattern data 40j from the ROM.

  In the first and third embodiments, the first estimation unit 251 selects one sound collection area (41jR, 41jL) from each of the left and right sound collection areas 411R to 414R, 411L to 414L, and further The second estimation unit 252 estimates in which of 41jR and 41jL the true sound source exists, but the second estimation unit is not necessarily provided.

  For example, when there is no noise source on the opposite side of the true sound source, such as when the teleconferencing device is used only on the right side or the left side, a synthesized signal (or difference) of both the sound collection areas 41jR and 41jL This is because there is no problem even if the signal is directly output to the counterpart device as a sound collection signal.

  The numerical values and the like shown in these embodiments do not limit the present invention. Further, in the above diagram, when there is a signal exchange between the configurations of the blocks that exhibit the functions, even in the configuration in which some of the functions of these blocks are processed by the other block, The same effect may be produced.

The figure which shows the external appearance and usage pattern of the remote conference apparatus which concerns on 1st Embodiment of this invention The figure explaining the sound beam and sound collection beam of the teleconference device The figure explaining the sound collection area set to the microphone array of the remote conference device Block diagram of the transmitter of the teleconference device Configuration diagram of first beam generation unit of the teleconference device Block diagram of the receiving unit of the teleconference device The block diagram of the transmission part of the remote conference apparatus of 2nd Embodiment of this invention The block diagram of the transmission part of the remote conference apparatus of 3rd Embodiment of this invention

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 ... Remote conference apparatus, 2 ... Transmission part, 22 ... Difference value calculation circuit, 231 ... 1st beam generation part, 232 ... 2nd beam generation part, 237 ... 3rd beam generation part, 238 ... 4th beam generation part, 239 ... Fifth beam generation unit, 251 ... First estimation unit, 252 ... Second estimation unit, 253 ... Third estimation unit, 254 ... Fourth estimation unit, 26 ... Signal selection unit, 271,272 ... Selector, 28 ... Multiplexer 3 ... receiver, 31 ... data receiver, 32 ... parameter calculator, 33 ... directivity controller 45 j (j = 1 to K) ... delay processor, 40 j (j = 1 to K) ... delay pattern Memory, 461i (i = 1 to N) ... Delay, 47j (j = 1 to K) ... Microphone input synthesis unit SPi (i = 1 to M) ... Speaker, SPA ... Speaker array ML, MR ... Microphone array, MLi ( i = 1 to N), MRi (i = 1 to 1) N) ... Microphone 100 ... Desk, 101 ... Center lines 411R to 414R, 411L to 414L ... Sound collection area, 999 ... Sound source (speaker)

Claims (1)

A speaker array comprising a plurality of speakers for outputting sound;
First and second microphone arrays provided to pick up sound on both sides in the longitudinal direction of the speaker array;
By synthesizing and synthesizing the audio signals picked up by the microphones of the first microphone array and the second microphone array by delay processing, a plurality of positions are symmetrically located with respect to the longitudinal center line of the speaker array. Sound collection area setting means for setting each of the first sound collection area and the plurality of second sound collection areas;
Differences for calculating differential signals of sound signals collected from the pair of sound collection areas at symmetrical positions among the sound signals collected from the plurality of first sound collection areas and the plurality of second sound collection areas, respectively. Signal calculation means;
First sound source position estimating means for selecting a sound collection area pair having a high signal intensity of the differential signal;
A sound collection area with a higher intensity of the collected sound signal is selected from the sound collection area pair selected by the first sound source position estimating means, and a second sound source position is estimated to be present in the sound collection area. Sound source position estimation means,
Equipped with a,
The sound collection area setting means further sets a plurality of narrow sound collection areas in the sound collection area selected by the second sound source position estimation means, and focuses each of the plurality of narrow sound collection areas. A function to generate a narrow sound collection beam of
A remote conferencing apparatus comprising: a third sound source position estimating means for estimating that a sound source position is in an area where the intensity of the collected sound signal is large among the plurality of narrow sound collection areas .

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