JP2002031674A - Method for correcting sounding body directivity and its apparatus - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide a method for correcting a sounding body directivity which prevents a high frequency region from being indistinct even when a gain in the rear of a speaker becomes high in a delay sum array method. SOLUTION: In the delay sum array method, a position of the sounding body is detected from output signals of a plurality of microphones. Output signals from a plurality of microphones near the sounding body (near microphone output signals) among the plurality of microphones are monitored by a frequency region. A direction of the microphone having the highest high region of a spectrum envelope of frequency characteristics among the near microphone output signals is detected as a front face of the sounding body. A difference from frequency characteristics of the other microphones is detected and corrected.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a method for processing the output signal of a microphone array composed of a plurality of microphones to collect a target sound with a high SN ratio, and thereby to determine the directivity of a sounding body such as a speaker. The present invention relates to a correction method and an apparatus therefor,
In particular, outputs from a plurality of microphones located close to the sounding body (hereinafter, referred to as “proximal microphone output”) are monitored in the frequency domain, and among the nearby microphone outputs, the highest band of the spectral envelope of the frequency characteristic is the highest. The present invention relates to a method and an apparatus for detecting a direction of a microphone that is raised as a front of a sounding body, detecting a difference from a frequency characteristic of another microphone, and correcting the difference.

[0002]

2. Description of the Related Art In recent years, with the advancement of multimedia technology, it has become possible to conduct a teleconference such as a video conference using a microphone and a speaker in a voice call mode.
In such a case, there is a need for a sound collection device that can make a natural call without being conscious of the microphone without installing microphones for the number of speakers on the desk of the communication conference and that collects only a target sound such as voice. .

[0003] As an example of such a sound pickup device, there is a sound pickup device in which a plurality of microphones (microphone arrays) are installed and their outputs are subjected to signal processing to extract a target sound. Signal processing methods for suppressing noise and extracting a target sound using such a microphone array include a delay-and-sum method and an AM method.
Although many are known, such as NOR (eg, Oga, Yamazaki, Kaneda, "Acoustic system and digital processing" IEICE, 1995, pp.173-197), for example, the purpose of the delay-sum method is as follows Extract sound.

FIG. 1 is a diagram for explaining the principle of target sound extraction by the delay-and-sum method. In Figure 1, 1 is sound pickup unit (microphone _{array), 2 1, 2 2,} ···, 2 M microphones (M is the number of _{microphones), 3 1, 3 2,} ···, 3 M delay , 4 is an adder, 5 is an output signal, 6 is a noise suppression unit,
d is a microphone interval, s (t) is a sound wave arriving at the sound pickup unit 1 (t represents time), and θ is an arrival angle at which the sound wave s (t) arrives at the sound pickup unit 1.

The microphones 2 ₁ , 2 ₂ ,..., 2 _{M in} FIG.
Are arranged linearly at equal intervals d, and the sound wave s (t) arrives at a microphone θ at an angle θ from a distance. The distance the sound wave reaching the microphone 2 ₁ propagates before reaching the microphone 2 ₂ is represented by dsinθ from the arrival angle θ a microphone spacing d (Figure 1). Similarly, the i-th microphone 2 _i (i = 2,..., M)
Is represented by (i−1) dsin θ. Accordingly, the microphone _{2 i (i = 2, ···} , M) is a delay time tau _i to reach, when the reference microphones 2 _1, by dividing the propagation distance at the speed of sound c, the following equation (1 )
It is represented by

Τ _i = (i−1) dsin θ / c (1) Here, when each microphone 2 _i (i = 1,..., M) represents an output signal X _i (t), this is a sound wave. Since s (t) is delayed by τ _i , the following equation (2) is obtained. _{X i (t) = s (} t-τ i) (2) where the delay unit _{3 i (i = 1, ···} , M) when appropriately setting the delay amount D _i of sound waves arriving from θ direction The following shows that the output signal 5 can be output with only the emphasis.

The delay amount D _i of the delay unit 3 _i (i = 1,..., M) is set as in the following equation (3). D _i = D ₀ −τ _i (3) Here, D ₀ is a fixed delay amount added in order to prevent a decrease in accuracy in realizing delay characteristics with a digital filter when the value of τ _i is too small. . At this time, the delay unit 3
_i (i = 1,..., M) is obtained by adding the signal of equation (2) to the delay D of equation (3).
_{Since i} occurs, the following equation (4) is obtained.

X _i (t−D _i ) = s (t−τ _i −D _i ) = s (t−τ _i − (D ₀ −τ _i )) = s (t−D ₀ ) (4) , Regardless of the microphone number i, s (t) is D ₀
The same signal is delayed. When the signals are added by the adder 4 after the phases are aligned in this way, the sound waves arriving from the θ direction are emphasized by the added amount.
On the other hand, sound waves and theta directions coming from another theta _N direction, tau _i
Since the sound is received with a delay time τ _N different from
With the delay amount of, the phases are not aligned, and the signals are not emphasized even when the signals are added by the adder 4.

As described above, in the delay-and-sum method, the sound wave coming from the target direction θ is emphasized, and the noise coming from the other direction θ _N is relatively suppressed. At this time, the desired direction θ
Is scanned, and the output signal of the microphone array is monitored, the output signal increases when θ is directed to the direction of the target speaker, so that the direction of the target speaker can be searched. Then, the target sound is increased by aligning and adding the phases according to Equation (4) so as to emphasize the sound wave from the direction θ of the target speaker, that is, by directing the directivity of the microphone array in the direction of θ. Sound can be collected at the SN ratio.

Here, for convenience of explanation, a plurality of microphones are described as being arranged on a straight line at equal intervals d. However, the microphones can be arranged at irregular intervals, and the arrangement shape is two-dimensional. Or three-dimensionally. When the point sound source S is located at a relatively short distance from the microphone array as shown in FIG. 2, the delay units 3 ₁ , 3 ₂ ,.・
It is important to improve the sound pickup S / N ratio by providing gain sections 7 ₁ , 7 ₂ ,..., 7 _M at the subsequent stage of 3 _M and applying an appropriate load (gain) to these gain sections. As a method of applying a load, there is a method of applying a load represented by the following equations (5), (6), and (7) (Nomura,
Kaneda, Kojima, “Near-field Microphone Array”, Journal of the Acoustical Society of Japan, 53, 2 (1997), pp. 110-116).

[0011]

(Equation 1) Here, r _1, r _2, · · ·, r _{M 1} 2 each microphone from the sound source S, ₂ 2, · · ·, distance to 2 _M, r _c is the critical length of the chamber, i.e., the direct sound of the sound source This is the distance at which the power and the reverberation sound power become equal, and the room volume V (m ³ ) and the room reverberation time T
(Sec), r _c = √ (0.0032V / T) (H.Kutt
ruff, `` Room Acoustics (Third Edition) '', Elsevier Ap
plied Science, pp. 100-132 (1991)). At this time, the microphone array has the highest sensitivity with respect to the "point" of the position of the sound source S, so that a "focus" of the sensitivity is formed. At this time, the distance r to each microphone
delay units 3 ₁ , 3 ₂ ,..., 3 _M for _i (i = 1,..., M)
Delay _{D 0 -r i / c (c} : sound velocity) and by changing the above-mentioned gain g ₀ i.e. a scan the focal point of the sensitivity, by monitoring an array output, the sensitivity to the point of presence of the target speaker When the focus is turned on, the array output becomes large, so that the position of the target speaker can be found.

In this way, by finding the target speaker's existence area as the direction or position and directing the array directivity to the existence area, the target sound can be picked up with a high sound collection SN ratio.

[0013]

When a human utters, its directivity is facing forward, and its high-frequency component is generally
Decays toward the rear (for example, "Hearing and Speech" edited by the Institute of Electronics and Communication Engineers, 1975, published by Corona, p. 236). This is shown in FIG. This figure shows the directional frequency characteristics of the human mouth with reference to the front. From this figure, it can be seen that the frequency is about 5 dB at 500 Hz and about 10 dB at 4 kHz with respect to the front. At this time, the delay-and-sum array method of FIG.
There is a problem that high-frequency muffled sound is picked up when the focus is directed to the vicinity of the back of the speaker's head.

[0014]

In order to solve the above-mentioned problems, a sound source position detecting means for detecting a position of a sounding body such as a speaker, and a plurality of microphone outputs close to the sounding body among outputs of the microphone. Frequency characteristic monitoring means for monitoring the output from the microphone (hereinafter referred to as “proximal microphone output”) in the frequency domain, and the microphone having the highest spectral envelope of the frequency characteristic among the nearby microphone outputs. There are provided a sounding body directivity detecting means for detecting the direction as the front of the sounding body, a spectrum difference detecting means for detecting a difference between the spectrum characteristics of other microphone outputs, and a spectrum difference correcting means for correcting the difference (see FIG. 4). .

According to the present invention, even if the calculation result of the weight gain in the delay-and-sum array method of FIG. 2 results in increasing the gain behind the speaker, the muffled sound in the high frequency range can be collected. The problem of being sounded can be avoided.

[0016]

Embodiments of the present invention will be described below with reference to the drawings. FIG. 5 shows a first embodiment of the present invention. In this figure, a microphone array 22 of a sounding body directivity correction device 21 is two-dimensionally (planarly) arranged on a ceiling or the like.
The output xi (t) of this microphone array 22 is
The delay unit 3 and the load unit 7 are scanned by the delay unit 3, and the phase is adjusted by the delay unit 3 for each focal point to become a signal yi (t).
Gi × yi (t). The output gi × yi (t) is sent to the power calculator 23, and the power (Σgi × yi (t)) ^ 2 is calculated. The sound source position detector 24 compares the powers of the respective focal points, and detects the focal position having the maximum power as the sound source position. The sound source position information is sent to the delay unit 3 ', and a signal y' (t) in which the phases of the microphone outputs xi (t) are aligned so that the focus is directed to the sound source position is sent to the front candidate extracting unit 25. Front candidate extraction unit 25
In (y), a microphone output (a signal having the same phase) near the sound source position is extracted from y (t) as a front candidate signal yi '(t) and sent to the high-pass filter 26, and yi other than yi' (t) (t) to the load 7 ". The signal yi" (t) passed through the high-pass filter 26 is sent.
Is sent to the power calculator 27, and the high-frequency power (yi "(t)) ^ 2
Is calculated and sent to the front direction detection unit 27. The front direction detection unit 27 outputs the microphone output (yi
(t)) is detected as the front direction, and the front direction information is sent to the front direction determination unit 30. On the other hand, the front candidate yi ′ (t) is subjected to FFT (Fast Fourier Transform) by the frequency domain transform unit to be Yi (ω), sent to the spectrum envelope extraction unit 29, and sent to the spectrum envelope Si
(ω) is calculated. The front direction determining unit 30 selects the front direction spectrum envelope S0 (ω) from Si (ω) based on the front direction information sent by the front direction determining unit 30. The spectrum difference detection unit 31 calculates Di (ω) = S0 (ω) / Si (ω) and selects the spectrum envelope S0 (ω). In the spectrum difference detector 31,
Di (ω) = S0 (ω) / Si (ω) is calculated, and the spectrum difference correction unit 32 calculates Zi (ω) = Yi (ω) × Di (ω) to perform spectrum correction. This Zi (ω) is inversely FFTed by the time domain conversion unit 33 and converted into a time waveform zi (t). This zi (t) is sent to the load unit 7 ′ having a load based on the sound source position information from the sound source position detection unit 24 and becomes gi × zi (t), and zi sent to the load unit 7 ″.
(t) becomes gi × yi (t), which is sent to the adder 4 ′, summed up, and sent to the output 5.

In the above embodiment, the microphone array may be arranged three-dimensionally instead of two-dimensionally (in a plane) on the ceiling or the like.

[0018]

As described above, according to the present invention, even if the calculation result of the weight gain in the delay-and-sum array method shown in FIG. Position detection means, frequency characteristic monitoring means for monitoring, in the frequency domain, outputs from a plurality of microphones located at a position close to the sounding body (hereinafter referred to as “proximal microphone output”), Of the microphone output, a sounding body directivity detecting means for detecting, as the front of the sounding body, the direction of the microphone in which the high frequency of the spectral envelope of the frequency characteristic is the highest, and the other microphone is detected by the spectrum difference detecting means. Since the difference from the frequency characteristic is detected and the difference is corrected by the spectrum difference correcting means, even if the gain behind the human becomes high by the delay-and-sum array method of FIG. That prevented, an excellent effect than ever before.

[Brief description of the drawings]

FIG. 1 is a view for explaining the principle of noise suppression sound pickup by a delay-and-sum method.

FIG. 2 is a diagram for explaining how to appropriately set a load of a gain at a subsequent stage of a delay unit and improve a sound collection S / N ratio when a sound source is located at a position close to a microphone array.

FIG. 3 is a diagram for explaining directivity of a human mouth.

FIG. 4 is a diagram for explaining how to detect the directivity of a sounding body.

FIG. 5 is a configuration diagram showing an embodiment of the present invention.

[Explanation of symbols]

 DESCRIPTION OF SYMBOLS 1 Sound collection part 2 Microphone 3 Delayer 4 Adder (Addition part) 6 Noise suppression part 7 Gain part (Load part) 11 Focus operation part 21 Sound emitting body directivity correction device 22 Microphone array 23 Power calculation part 24 Sound source position detection part 25 Front candidate extraction unit 26 High pass filter 27 Front direction detection unit 28 Frequency domain conversion unit 29 Spectrum envelope extraction unit 30 Front direction determination unit 31 Spectrum difference detection unit 32 Spectrum difference correction unit 33 Time domain conversion unit

Claims (4)

[Claims] 1. A sound source directivity correction method in a sound collection method using a microphone array including a plurality of microphones and a microphone array device for processing an output signal of the microphone array, wherein sound is generated from output signals of the plurality of microphones. The body position is detected, and among the output signals of the microphone, output signals from a plurality of microphones close to the sounding body (hereinafter, referred to as “proximal microphone output signal”) are monitored in a frequency domain, and the output of the nearby microphone is monitored. Detecting the direction of the microphone where the high frequency of the spectral envelope of the frequency characteristic is the highest in the signal as the front of the sounding body, detecting the difference from the frequency characteristics of other microphones, and correcting the difference A method for correcting directivity of a sounding body, characterized by the following. 2. A sounding body directivity correction device for a sound pickup device using a microphone array comprising a plurality of microphones and a microphone array device for processing an output signal of the microphone array, wherein output signals of the plurality of microphones are inputted. Sound source position detecting means for detecting the position of the sounding body, and output signals from a plurality of microphones located closer to the sounding body among the output signals of the microphone (hereinafter, referred to as "proximal microphone output signal")
Characteristic monitoring means for monitoring the sound in the frequency domain, and sounding body directivity for detecting, as the front of the sounding body, the direction of the microphone having the highest frequency envelope of the frequency characteristic of the nearby microphone output signal. A sounding body directivity correction apparatus, comprising: a spectrum difference detection means for detecting a difference between the detection means and a spectrum characteristic of another microphone output signal; and a spectrum difference correction means for correcting the difference. 3. The sounding body directivity correction apparatus according to claim 2, wherein said frequency characteristic monitoring means includes a frequency domain conversion means for converting a nearby microphone output signal into a frequency domain, and a spectrum envelope of a frequency characteristic from the frequency domain signal. A sound envelop directivity correction device comprising a spectrum envelope extracting means for extracting the sound. 4. The sounding body directivity correction apparatus according to claim 3, wherein the spectrum envelope extracting means extracts a low-order cepstrum from the frequency domain signal, and extracts the spectrum envelope from the low-order cepstrum. A sounding body directivity correction apparatus, characterized by comprising extraction means.