JP2009124454A - Echo elimination method, device, program, and recording medium - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To suppress even distortion components caused by a speaker in echo components in a microphone sound reception signal. <P>SOLUTION: A coupling amount is estimated based on level ratio between a frequency region main microphone sound receiving signal and a frequency region sub-microphone sound receiving signal by an echo estimation part 20, an estimated echo level is obtained by multiplication between the estimated coupling amount and the frequency region sub-microphone sound receiving signal, ratio between difference between the level of the frequency region main microphone sound receiving signal and the level of the estimated echo level and the level of the frequency region main microphone sound receiving signal is calculated as a gain by an echo suppression gain calculation part 14, the gain is multiplied by the frequency region main microphone sound receiving signal to suppress the echo components in the frequency region main microphone sound receiving signal by a multiplication part 16, and inverse frequency region conversion is performed to a multiplication result of the gain by an inverse frequency region conversion part to output a transmission signal of a time region. <P>COPYRIGHT: (C)2009,JPO&INPIT

Description

  The present invention relates to an echo cancellation method and apparatus for hands-free communication such as a video conference and an audio conference.

A conventional echo canceller will be described.
FIG. 9 is a block diagram of a conventional echo canceller disclosed in Patent Document 1. In FIG. The prior art echo canceller 10 includes a frequency domain converter 11, a coupling amount estimator 12, an echo level estimator 13, an echo suppression gain calculator 14, a frequency domain converter 15, and a gain multiplier 16. , And the inverse frequency domain transform unit 17.

  This echo canceller erases acoustic echo, which is speaker sound mixed in a microphone sound reception signal in a loudspeaking call using a speaker and a microphone. The input signal of the echo canceller 10 is a received signal x (t) which is a signal received from the other party of the call and output as a received voice from the speaker 31 at the local point, and a microphone which is a signal received by the microphone 32 at the local point. The received signal is y (t), and the output signal is a transmission signal e (t) that is a transmission signal to the other party. This device 10 estimates the frequency spectrum of the acoustic echo included in the microphone received signal y (t), obtains a gain for suppressing the acoustic echo in the microphone received signal, and obtains the microphone received signal converted into the frequency domain. Is multiplied by a gain for suppressing the acoustic echo to cancel the acoustic echo contained in the microphone sound reception signal.

Hereinafter, the operation of the conventional echo canceling apparatus will be described with reference to FIG.
FIG. 10 shows the relationship between signals in the echo canceller 10, the speaker 31, and the microphone 32 in FIG. 9 as an equivalent model. A signal obtained by converting the received signal x (t), microphone received signal y (t), and transmitted signal e (t) into the frequency domain in a short-time frame, respectively, X (ω, n), Y (ω, n) , E (ω, n). However, discretized t represents time, ω represents a discretized frequency, and n represents a frame number. Here, the conversion procedure to the frequency domain will be described in detail with reference to FIG. 11 by taking the frequency domain conversion of the received signal x (t) as an example. The N-sample signals from 0 to (N−1) th of the received signal x (t) are multiplied by a window function and then subjected to Fourier transform to obtain the frequency domain signal X (ω, 0). Next, the signal of N samples from T to (N−1 + T) th shifted by T samples is similarly multiplied by a window function and then subjected to Fourier transform to obtain the frequency domain signal X ( ω, 1) is obtained. This procedure is repeated while shifting by T samples, and the frequency domain signal X (ω, n) (n = 0, 1,...) In each frame is obtained.

  Next, in FIG. 9, the transmission characteristic of the sound transmission path 33 from the speaker 31 to the microphone 32 is linear, the spatial response (impulse response) is r (t), and the transfer function (impulse response is frequency domain transformed). ) Is represented by R (ω). In general, the speaker characteristic has a non-linear response characteristic such that the output reaches a peak with respect to the input of a signal having a large amplitude. Therefore, the characteristic of the speaker 31 in FIG. Considering this, the impulse response of the linear response unit 31A is expressed as g (t), the transfer function is expressed as G (ω), the characteristic in the time domain of the nonlinear response unit 31B is expressed as a function f, and the characteristic in the frequency domain is expressed as F. The outputs of the linear response unit 31A and the non-linear response unit 31B are added by the addition unit 31C to be the output of the speaker 31. Also, s (t) is the voice of the near-end speaker 40 and S (ω, n) is the near-end talker voice converted to the frequency domain in a short frame, and the transmission path from the near-end talker 40 to the microphone 32 The impulse response of 34 is represented by c (t), and the transfer function is represented by C (ω).

From these, the microphone sound reception signal y (t) is expressed by the following equation (1).
y (t) = g (t) ^* r (t) ^* x (t) + r (t) ^* f (x (t)) + c (t) ^* s (t) (1)
However, ^* represents a convolution operation. If this is expressed in the frequency domain, Equation (2) is obtained.
Y (ω, n) ＝ G (ω) R (ω) X (ω, n) + R (ω) F (X (ω, n)) + C (ω) S (ω, n) (2)
Furthermore, when expressed in terms of a power spectrum, equation (3) is obtained.
| Y (ω, n) | ² = | G (ω) R (ω) | ² | X (ω, n) | ² + | R (ω) | ² | F (X (ω, n)) | ² ＋ | C (ω) | ² | S (ω, n) | ²
(3)
However, | · | represents an absolute value. Note that the first and second terms on the right side of Equation (3) correspond to the linear acoustic echo component and the nonlinear acoustic echo component from the speaker 31 to the microphone 32, respectively.

Here, the echo canceller is required to obtain a gain for suppressing the acoustic echo component included in the power spectrum | Y (ω) | ² of the microphone sound reception signal, and to calculate the gain in the frequency domain signal of the microphone sound reception signal. Multiplying Y (ω) to suppress the acoustic echo component. That is, the result of multiplying the power spectrum of the microphone sound reception signal y (t) by | Y (ω) | ² and the power spectrum H (ω, n) ² of the gain is the microphone 32 of the speaker voice s (t). Ideally, it becomes equal to the power spectrum of the input component to | C (ω) | ² | S (ω, n) | ² . If this is expressed by an equation, equation (4) is obtained.
H (ω, n) ² | Y (ω, n) | ² = | C (ω) | ² | S (ω, n) | ² (4)
Solving equation (4) for gain H (ω, n) yields equation (5).
H (ω, n) ＝ SQRT (| C (ω) | ² | S (ω, n) | ² / | Y (ω, n) | ² ) (5)
However, SQRT (•) means taking the square root.

しかし、図９に示す従来のエコー消去装置では、スピーカの非線形成分（即ち歪成分）である|F(X(ω,n))|²は観測不可能であるので、従来のエコー消去装置のエコー抑圧ゲイン計算部１４で計算されるゲイン値H_conv(ω,n)は、スピーカの歪成分を無視した次式(7)

により求めていた。このため、従来のエコー消去装置では、スピーカの歪成分の抑圧はできない。ここで、|G’(ω)R’(ω)|は、スピーカ３１からマイクロホン３２までの伝達関数の推定スペクトルであり推定結合量と呼ぶ。 Here, | C (ω) | ² | S (ω, n) | ² is not actually observable, so if equation (3) is modified and substituted into equation (5), It becomes like (6).

However, in the conventional echo canceller shown in FIG. 9, the non-linear component (ie, distortion component) of the speaker | F (X (ω, n)) | ² cannot be observed. The gain value H _conv (ω, n) calculated by the echo suppression gain calculation unit 14 is expressed by the following equation (7) ignoring the distortion component of the speaker.

Was asking for. For this reason, the conventional echo canceller cannot suppress the distortion component of the speaker. Here, | G ′ (ω) R ′ (ω) | is an estimated spectrum of a transfer function from the speaker 31 to the microphone 32 and is called an estimated coupling amount.

Thus far, the gain value calculated by the conventional echo canceller has been derived. Hereinafter, specific processing contents will be described with reference to FIG.
The frequency domain transform unit 11 transforms the received signal x (t) into a frequency domain signal X (ω, n) by the same processing as described in FIG. The frequency domain converter 15 converts the microphone sound reception signal y (t) into a frequency domain signal Y (ω, n). The conversion method is the same as that of the frequency domain converter 11.

The coupling amount estimation unit 12 estimates the coupling amount between the speaker 31 and the microphone 32 from the output X (ω, n) of the frequency domain conversion unit 11 and the output Y (ω, n) of the frequency domain conversion unit 15. G ′ (ω) R ′ (ω) | The amount of coupling is the amplitude of the transfer function between the speaker 31 and the microphone 32, and is determined by the ratio of the absolute value of the speaker output signal X (ω, n) in the frequency domain and the microphone sound reception signal Y (ω, n) in the frequency domain. It is done. In addition, time smoothing is performed to improve the accuracy of the coupling amount. That is, the estimated coupling amount | G ′ (ω) R ′ (ω) | can be obtained by the equation (8).
| G '(ω) R' (ω) | ＝ Avg {| Y (ω, n) | / | X (ω, n) |} (8)
However, Avg (·) represents taking an average value at each value of ω.

The echo level estimation unit 13 calculates the estimated echo from the output X (ω, n) of the frequency domain conversion unit 11 and the estimated coupling amount | G ′ (ω) R ′ (ω) | that is the output of the coupling amount estimation unit 12. Find the level | Z (ω, n) |. The estimated echo level is given by
| Z (ω, n) | ＝ | G '(ω) R' (ω) || X (ω, n) | (9)
As shown by multiplying the received signal by the coupling amount.

The echo suppression gain calculation unit 14 uses the gain H _conv (ω, n) to suppress the echo from the output | Z (ω, n) | of the echo level estimation unit 13 and the output Y (ω, n) of the frequency domain conversion unit 15. Find n). Since the equation for obtaining the gain has already been derived as the equation (7), it is calculated by the following equation (10) obtained by substituting the equation (9) into the equation (7).
H _conv (ω, n) = SQRT {(| Y (ω, n) | ² − | Z (ω, n) | ² ) / | Y (ω, n) | ² } (10)

The gain multiplication unit 16 multiplies the output Y (ω, n) of the frequency domain conversion unit 15 by the gain H _conv (ω, n) obtained by the echo suppression gain calculation unit 14 to obtain the frequency domain signal of the transmission signal. Find E (ω, n). If this is expressed by an equation, equation (11) is obtained.
E (ω, n) ＝ H _conv (ω, n) Y (ω, n) (11)

  The inverse frequency domain transform unit 17 performs inverse frequency domain transform on the output signal E (ω, n) of the gain multiplier 16 and outputs a transmission signal e (t) that is an output of the echo canceller 10. A detailed procedure of the inverse frequency domain transform will be described with reference to FIG. The output signal E (ω, n) of the gain multiplier 16 is subjected to inverse Fourier transform every N sample frames, and then multiplied by a window function. Next, the output signal e (t) is obtained and output by shifting the N sample signals obtained by multiplying these window functions by T samples.

By the method described above, the echo canceling device of the prior art cancels the echo component. However, in the conventional echo canceller, when the distortion of the speaker is large, the distortion component (nonlinear component) is not erased. This is because the distortion component | R (ω) | ² | F (ω, n)) | ² of the unobservable speaker in Equation (6) was ignored when deriving the equation to calculate the gain for suppressing echo. It is.
JP2005-250266

  In the conventional echo canceller, the echo that can be canceled is only the echo component that reaches the microphone through the linear echo path, and the nonlinear echo component cannot be canceled. Therefore, when a loudspeaker or the like with strong nonlinearity is used, there arises a problem that sufficient echo suppression performance cannot be obtained.

  The problem to be solved by the present invention is to provide an echo cancellation method and apparatus that realizes high echo cancellation performance even when a loudspeaker or the like with strong nonlinearity is used.

According to the present invention, an echo canceller for canceling an echo component, which is a signal component output from a speaker and circulated into a microphone, from a microphone sound reception signal,
A frequency domain converting means for converting a main microphone received signal and a sub microphone received signal obtained from a plurality of microphone received signals into a frequency domain main microphone received signal and a frequency domain sub microphone received signal, respectively;
The amount of coupling is estimated based on the level ratio between the frequency domain main microphone received signal and the frequency domain sub microphone received signal, and the estimated echo level is calculated by multiplying the estimated amount of coupling and the frequency domain sub microphone received signal. Echo estimation means for obtaining
An echo suppression gain calculating means for calculating a difference between the level of the frequency domain main microphone received signal and the estimated echo level and a ratio of the level of the frequency domain main microphone received signal as a gain;
A gain multiplier that multiplies the frequency domain main microphone received signal by the gain to suppress echo components in the frequency domain main microphone received signal;
An inverse frequency domain transform unit for performing a reverse frequency domain transform on the multiplication result by the gain multiplier and outputting a time domain transmission signal;
And is configured to include.

According to the present invention, an echo canceling method for canceling an echo component, which is a percussion synthesis component that is output from a speaker and circulates into a microphone, from the microphone received signal,
(a) a process of converting a main microphone received signal and a low sub microphone received signal obtained from a plurality of microphone received signals into a frequency domain main microphone received signal and a frequency domain sub microphone received signal, respectively;
(b) Estimating a coupling amount based on a level ratio between the frequency domain main microphone received signal and the frequency domain sub microphone received signal, and multiplying the estimated coupling amount by the frequency domain sub microphone received signal. The process of obtaining the estimated echo level;
(c) calculating as a gain the ratio between the level of the frequency domain main microphone received signal and the estimated echo level and the level of the frequency domain main microphone received signal;
(d) a process of suppressing an echo component in the frequency domain main microphone received signal by multiplying the frequency domain main microphone received signal by the gain;
(e) a process of outputting a time-domain transmission signal by performing inverse frequency domain transformation on the multiplication result obtained in the process (d),
Including.

  According to the present invention, it is possible to suppress both a linear acoustic echo and a non-linear acoustic echo generated due to the distortion characteristics of the speaker, thereby realizing a high echo cancellation performance. Further, high-quality sound collection can be realized with little deterioration of the near-end speaker voice.

Example 1
FIG. 1 is a block diagram of an echo canceller that is a first embodiment of the present invention. Corresponding parts in FIG. 9 are shown with similar reference numerals.
The echo canceller of the present embodiment, when the microphone 32 and the main microphones 32 ₁ and the frequency domain conversion unit 15 ₁ a which corresponds to the frequency domain transforming section 15 in the conventional arrangement of FIG. 9, the sub-microphone 32 ₂ to 32 _M ( M is an integer greater than or equal to ₂ ) and frequency domain transforming units 15 ₂ to 15 _M corresponding thereto, coupling amount estimating units 21 _{2 to} 21 _M , and echo level estimating units 22 _{2 to} 22 _M are added. The frequency domain conversion unit 11, the coupling amount estimation unit 12, and the echo level estimation unit 13 for the signal x (t) are omitted. Further, an echo level integration unit 23 that integrates the outputs of 22 _{2 to} 22 _M is provided, and the output is provided to the echo suppression gain calculation unit 14. The coupling amount estimation units 21 _{2 to} 21 _M , the echo level estimation units 22 _{2 to} 22 _M, and the echo level integration unit 23 constitute an echo estimation unit 20. However, as will be apparent from the description below, the echo level integration unit 23 is necessary when the number of microphones to be used is three or more, and is not necessary when the number is two.

FIG. 2 is a diagram for explaining input / output signals of the echo canceling apparatus 100 of this embodiment. The input signal of the echo canceling apparatus 100 according to the present embodiment includes the received voice output from the speaker 31 at the local point according to the received signal x (t) given from the other party of the call through the communication network 50, and the near-end speaker 40. And two or more microphone sound reception signals y ₁ (t) to y _M (t), which are signals received by two or more microphones 32 _{1 to} 32 _M at the local point, and the output signal is , A transmission signal e (t) which is a transmission signal to the other party. However, FIG. 2 shows a case where _two microphones 32 ₁ and 32 ₂ are used. Each microphone sound reception signal y _m (t) (m = 1, 2) is output from the speaker 31 in addition to the transmitted voice input from the near-end speaker 40 through the path 34 _m to the microphone 32 _m. Includes an acoustic echo component, which is a signal that has passed through the path 33 _m and has entered the microphone 32 _m . The echo canceling apparatus 100 cancels this acoustic echo component to facilitate conversation. Each input signal of the echo canceling apparatus 100 is converted from an analog signal to a discrete time signal by AD conversion, and each output signal is converted from a discrete time signal to an analog signal by DA conversion.

FIG. 3 equivalently shows the signal relationship among the echo canceller 100, the speaker 31, and the microphones 32 _{1 to} 32 _{M in} FIG. However, FIG. 3 shows a case where M = 2. The received signal is x (t), the M-channel microphone received signals received by two or more microphones 32 _{1 to} 32 _M are y ₁ (t),..., Y _M (t), and the transmitted signal is e (t ). Signals obtained by converting these signals into the frequency domain for each frame are represented as X (ω, n), Y ₁ (ω, n),..., Y _M (ω, n), and E (ω, n), respectively. The conversion method to the frequency domain is the same as that of the conventional echo canceller, and the conversion to the frequency domain is performed according to the procedure described in FIG. However, t represents a discrete time, ω represents a discrete frequency, and n represents a frame number.

Next, the transmission characteristics of the sound transmission paths 33 _{1 to} 33 _M from the speaker 31 to the microphones 32 _{1 to} 32 _M are linear, and their impulse responses are expressed as r ₁ (t),..., R _M (t), The transfer function (the impulse response converted into the frequency domain) is represented by R ₁ (ω),..., R _M (ω).

Since the speaker 31 has a nonlinear characteristic such that the output reaches a peak with respect to an input of a signal having a large amplitude, it has a linear response unit 31A having a linear response characteristic and a nonlinear response characteristic as in the case of FIG. Considering the nonlinear response unit 31B separately, the impulse response of the linear response unit 31A is g (t), the transfer function is G (ω), the time domain characteristic of the nonlinear response unit 31B is the function f, and in the frequency domain The characteristic is expressed by F. S (t) the speech of the near-end talker 40, the frequency domain signal S (ω, n) and, from the near end talker 40 of the transmission path 34 ₁ to 34C _M to the microphones 32 ₁ to 32 _M an impulse response _{c 1 (t), ...,} c M (t), a transfer function C ₁ (ω), ..., represented by C _M (ω).

From these, the microphone sound reception signal y _m (t) can be expressed by the following equation (12). Here, m (m = 1,..., M) represents a microphone number.
y _m (t) = g (t) ^* r _m (t) ^* x (t) + r _m (t) ^* f (x (t)) + _cm (t) ^* s (t) (12)
Here, ^* represents a convolution operation. Each microphone sound reception signal Y _m (ω) after the frequency domain conversion is expressed by the following equation (13).
Y _m (ω, n) = G (ω) R _m (ω) X (ω, n) + R _m (ω) F (X (ω, n)) + C _m (ω) S (ω, n) (13 )

Further, the power spectrum | Y _m (ω) | ² of the microphone sound reception signal is expressed by the following equation (14).
| Y _m (ω, n) | ² = | G (ω) R _m (ω) | ² | X (ω, n) | ² + | R _m (ω) | ² | F (X (ω, n) ) | ²
＋ | C _m (ω) | ² | S (ω, n) | ² (14)
However, | · | represents an absolute value. The component of the first item on the right side is called a linear echo component, and the component of the second item on the right side is called a nonlinear echo component.

Here, the echo canceller is required to obtain a gain for suppressing the acoustic echo component included in the power spectrum | Y ₁ (ω) | ² of the main microphone sound reception signal y ₁ (t), The acoustic echo component is suppressed by multiplying the frequency domain signal Y ₁ (ω) of the microphone sound reception signal y ₁ (t). That is, when the power spectrum | Y ₁ (ω) | ² of the main microphone sound reception signal y ₁ (t) is multiplied by the power spectrum H (ω, n) ² of the gain, the main microphone 32 of the speaker voice is obtained. Ideally, the power spectrum of the input speech component to ₁ becomes | C _m (ω) | ² | S (ω, n) | ² . If this is expressed by an equation, equation (15) is obtained.
H (ω, n) ² | Y ₁ (ω, n) | ² = | C ₁ (ω) | ² | S (ω, n) | ² (15)
Solving equation (15) for gain H (ω, n) yields equation (16).
H (ω, n) ＝ SQRT {| C ₁ (ω) | ² | S (ω, n) | ² / | Y ₁ (ω, n) | ² } (16)
However, SQRT (•) means taking the square root. Here, | C ₁ (ω) | ² | S (ω, n) | ² is not actually observable, so if equation (14) is transformed into equation (16), then Equation (17) is obtained.

In order to obtain the gain represented by Equation (17), the power spectrum of the echo signal component obtained by adding the linear echo component and the nonlinear echo component (| G (ω) R ₁ (ω) | ² | X (ω , n) | ² + | R ₁ (ω) | ² | F (X (ω, n)) | ² ) must be estimated. _{Y m (ω, n) |} | power spectrum of the sub-microphone sound reception signal ^{_{y m (t) 2, m}} = 2, ..., the sub-microphone 32 _m and the main microphone 32 obtained by M, the coupling amount estimating unit 12 _m binding of the estimated value of between _{1 | R '1 (ω)} | / | R' m (ω) | power spectrum _{| R '1 (ω) |} 2 / | R' m (ω) | 2 the echo level If multiplication is performed by the estimation unit 13 _m , the following equation (18) is obtained using equation (14).
| Y _m (ω, n) | ² | R ' ₁ (ω) | ² / | R' _m (ω) | ²
= | G (ω) | ² | R ' ₁ (ω) | ² (| R _m (ω) | ² / | R' _m (ω) | ² ) | X (ω, n) | ²
+ | R ' ₁ (ω) | ² (| R _m (ω) | ² / | R' _m (ω) | ² ) | F (X (ω, n)) | ²
+ (| R ' ₁ (ω) | ² / | R' _m (ω) | ² ) | C _m (ω) | ² | S (ω, n) | ²
≒ | G (ω) | ² | R ' ₁ (ω) | ² | X (ω, n) | ² + | R' ₁ (ω) | ² | F (X (ω, n)) | ²
+ (| R ' ₁ (ω) | ² / | R' _m (ω) | ² ) | C _m (ω) | ² | S (ω, n) | ² (18)
Is obtained. However, | R _m (ω) | ² / | R ′ _m (ω) | ² = 1. (| R ' ₁ (ω) | ² / | R' _m (ω) | ² ) | C _m (ω) | If ² is sufficiently small, the sum of the estimated linear echo component and the estimated nonlinear echo component is obtained from equation (18). The power spectrum of the estimated echo signal is
| Z _m (ω, n) | ² = | Y _m (ω, n) | ² | R ' ₁ (ω) | ² / | R' _m (ω) | ²
≒ | G (ω) R ' ₁ (ω) | ² | X (ω, n) | ² + | R' ₁ (ω) | ² | F (X (ω, n)) | ² (19)
It can be seen that In order for this (| R ' ₁ (ω) | ² / | R' _m (ω) | ² ) | C _m (ω) | ^{2 to} be sufficiently small, the arrangement of microphones and speakers must be devised. This is necessary and will be described later.

In the configuration of FIG. 1, the number of sub-microphone one (i.e. M = 2) even better, but to increase the accuracy of the approximation, the number of sub microphones 2 or more, each of the sub and main microphone 32 ₁ as described below the amount of coupling between the microphones _{32 m | R '1 (ω} ) | / | R' m (ω) | Z m determined from the estimated value (ω, n), m = 2, ..., integrates M, a Estimated echo level Z (ω, n) is determined.

Next, the estimated coupling amount | R '(ω) | / | R' ₂ (ω) | to | R '(, respectively, for the received sound signal | Y ₂ (ω) | to | Y _M (ω) | ω) | / | R ' _M (ω) | multiplied by the estimated echo level Z ₂ (ω, n) -Z _M (ω, n) 17) Sum of power spectrum of linear echo component and nonlinear echo component in | G (ω) R ₁ (ω) | ² | X (ω, n) | ² + | R ₁ (ω) | ² | F (X (ω, n)) | ² is estimated. As a method for integrating the estimated echo levels, there are a method for obtaining the maximum value and an average method for each value of ω of the obtained estimated echo levels Z ₂ (ω, n) to Z _M (ω, n).

When the estimated echo level after integration is expressed as Z (ω, n), the echo suppression gain calculated in this embodiment is expressed by the following equation (20)
H _prop (ω, n) ＝ SQRT {(| Y (ω, n) | ² − | Z (ω, n) | ² ) / | Y (ω, n) | ² } (20)
It is represented by Here, since the estimated echo level Z (ω, n) of the present embodiment includes both components of the linear echo and the nonlinear echo, both of them can be suppressed.

Up to this point, an expression for obtaining the gain H _prop (ω, n) for suppressing the echo in this embodiment has been derived. In equation (20), the gain is calculated based on the signal power level, but the gain may be calculated based on the amplitude level of the signal as in equation (20 ′) below.
H _prop (ω, n) = (| Y (ω, n) | − | Z (ω, n) |) / | Y (ω, n) | (20 ')
According to the equation (20 ′), the gain accuracy is deteriorated, but there is an advantage that the calculation amount is reduced. The same applies to other embodiments described later, and when calculating the gain, the level of the signal to be handled may be a power level or an amplitude level.

The specific operation of each part in FIG. 1 will be described below.
Frequency domain transform section 15 ₂ to 15 _M for the frequency domain transform section ₁₅₁ and the sub microphone received sound signals to the main microphone sound reception signal, the microphone sound reception signal _{y 1 (t) ~y M (} t) a frequency domain signal Convert from Y ₁ (ω, n) to Y _M (ω, n). The conversion method is the same as the frequency domain conversion method for the received signal x (t) described in FIG.

Sub microphone sound reception signal _{y 2 (t) ~y M (} t) coupling amount estimating section 12 ₂ to 12 _M provided for each includes a respective sub-microphone received sound signal _{y 2 (t) ~y M (} t) The amount of coupling between the main microphone sound reception signal y ₁ (t) is estimated. The amount of coupling is the amplitude of the transfer function between the speaker and the microphone, and is determined by the ratio of absolute values in the frequency domain of the speaker output signal and the microphone sound reception signal. In addition, time smoothing is performed to improve the accuracy of the coupling amount. For example, the estimated coupling amount | G ′ (ω) R ′ ₁ (ω) | is obtained by Expression (21).
| G '(ω) R' ₁ (ω) | ＝ Avg {| Y ₁ (ω, n) | / | X (ω, n) |} (21)
However, Avg (·) represents taking an average value at each value of ω.

Next, coupling estimation unit 12 m for the sub-microphone received sound signal _{y m (t) (m =} 2, ..., M) , the sub-microphone received sound signal y _m (t) with respect to the frequency domain converter 15 _m outputs of Y _m (ω, n) and, from the main microphone sound reception signal y ₁ output Y ₁ of the frequency domain transform unit 15 ₁ for (t) (ω, n) , the amount of coupling between the main microphone 32 ₁ and the sub microphone 32 _m Estimated value | R ′ ₁ (ω) | / | R ′ _m (ω) | In the same manner as the coupling amount estimation unit 12 for the received signal x (t), the coupling amount | R ′ ₁ (ω) | / | R ′ _m (ω) | is estimated by Expression (22).
| R ' ₁ (ω) | / | R' _m (ω) | ＝ Avg {| Y ₁ (ω, n) | / | Y _m (ω, n) |} (22)
However, Avg (·) represents taking an average value at each value of ω.

The echo level estimation unit 13 _m for the sub microphone sound reception signal y _m (t), m = 2,..., M is the output Y _m of the frequency domain conversion unit 15 _m for the sub microphone sound reception signal y _m (t). From (ω, n), m = 2,..., M and the output | R ′ ₁ (ω) | / | R ′ _m (ω) | of the coupling amount estimation unit 12 _m , the estimated echo level | Z _m (ω , n) |, m = 2,. This estimated echo level is obtained by multiplying the sub-microphone sound reception signal and the coupling amount as shown in the following equation (23).
| Z _m (ω, n) | ＝ (| R ' ₁ (ω) | / | R' _m (ω) |) | Y _m (ω, n) | (23)

Echo level integration unit 18, the output of the echo level estimator _{13 2 ~13 M | Z m (} ω, n) |, m = 2, ..., a M, 1 single echo level | Z (ω, n) | To integrate. Integration is performed by a method of taking a maximum value or a method of taking an average value. In the method of taking the maximum value, the integrated echo level | Z (ω, n) | is obtained by the following equation (24), and in the method of taking the average value, the integrated echo level | Z (ω, n ) |
| Z (ω, n) | ＝ Max _m (| Z _m (ω, n) |) (24)
| Z (ω, n) | ＝ Ave _m (| Z _m (ω, n) |) (25)
However, Max _m (•) means that the maximum value is obtained by changing m at each value of ω, and Ave _m (•) means that the average value is obtained by changing m at each value of ω. To do.

Echo suppression gain calculator 14, the output of the echo level of integration unit 18 | Z (ω, n) | and the main microphone sound reception signal y ₁ output Y ₁ of the frequency domain transform unit 15 ₁ for (t) (ω, n ), The gain H _prop (ω, n) for suppressing the echo is obtained. Gain was obtained by subtracting the integrated echo-level power spectrum | Z (ω, n) | ² from the power spectrum | Y ₁ (ω, n) | ² of the main microphone sound reception signal y ₁ (t) The estimated transmission voice level is obtained by dividing by the power spectrum | Y ₁ (ω, n) | ² of the main microphone sound reception signal y ₁ (t). This can be expressed by the following equation (26).
H _conv (ω, n) = SQRT {(| Y ₁ (ω, n) | ² − | Z (ω, n) | ² ) / | Y ₁ (ω, n) | ² } (26)

Gain multiplication unit 16, the gain H _prop (ω, n) obtained in the echo suppression gain computing unit 14, a main microphone sound reception signal y ₁ output Y ₁ of the frequency domain transform unit 15 ₁ for (t) (omega, By multiplying n), the frequency domain signal E (ω, n) of the transmission signal e (t) is obtained. If this is expressed by an equation, equation (27) is obtained.
E (ω, n) ＝ H _prop (ω, n) Y ₁ (ω, n) (27)

  The inverse frequency domain transform unit 17 performs inverse frequency domain transform on the output signal E (ω, n) of the gain multiplication unit 16 by the same process as described with reference to FIG. 12, and outputs the output of the echo cancellation apparatus 100 of the present embodiment. The transmission signal e (t) is output.

Next, the arrangement of microphones and speakers will be described.
The estimated echo level | Z _m (ω, n) | obtained from the sub microphone received signal y _m (t) (m = 2,…, M) has a near-end talk as shown in Equation (18). Human voice component | S (ω, n) | is also mixed. If the estimated echo level includes a component of the near-end speaker voice, a gain that suppresses even the near-end speaker voice is determined, so that the transmitted voice is deteriorated. In order to prevent this, it is necessary to reduce (| R ′ ₁ (ω) | ² / | R ′ _m (ω) | ² ) | C _m (ω) | ² in equation (18). For this purpose, the amplitude of the transfer function from the speaker 31 to the sub microphone 32 _m | R ′ _m (ω) |, m = 2,..., M is large, and the amplitude of the transfer function from the speaker 31 to the main microphone 32 ₁ | R ′ ₁ (ω) | is small, and the amplitude | C _m (ω) |, m = 2,..., M of the transfer function from the near-end speaker to the sub microphone 32 _m may be small. One possible method is to devise a microphone arrangement. For example, as shown in FIG. 4, using two unidirectional microphone, the main microphones 32 ₁ directs the sensitive direction near-end talker 40, directs the low sensitivity direction to the speaker 31. The sub-microphone 32 _2, conversely toward the sensitive direction to the speaker 31, directs a low sensitivity direction near-end talker 40. With such an _{arrangement, | C m (ω) |} , m = 2, ..., the amplitude of M is reduced, _{further, | R '1 (ω)} | amplitude becomes small, | R' _m The amplitude of (ω) |, m = 2,. By this device, it is possible to reduce the deterioration of the near-end speaker voice.

  As described above, in the present invention, the echo estimation unit 20 estimates the echo level based on the coupling amount between the main microphone received signal and the sub microphone received signal. As shown in Equation (18), the estimated echo level includes a linear echo component and a nonlinear echo component. Therefore, the echo suppression gain obtained from such an estimated echo level is multiplied by the main microphone sound reception signal. As a result, both the linear echo component and the nonlinear echo component included in the main microphone sound reception signal can be suppressed.

  The echo canceling apparatus 100 according to the present embodiment can realize a comfortable hands-free call by suppressing both nonlinear echoes and linear echo components caused by the distortion components even when the speaker distortion is large.

In the embodiment of FIG. 1, M is an integer of 2 or more, of course echo level integration section 23 in the case of M = 2 is not required, the output estimated echo level of the echo level estimating unit 22 _M Z _M (ω, n) is directly given to the echo suppression gain calculator 14.
Example 2

FIG. 5 is a block diagram showing an echo canceling apparatus according to a second embodiment of the present invention.
This embodiment is intended to further improve the echo estimation accuracy by combining the conventional echo estimation method described in FIG. 9 with the embodiment of FIG. Similarly, a frequency domain conversion unit 11, a coupling amount estimation unit 12, and an echo level estimation unit 13 for the received signal x (t) are added. The echo estimation unit 20 includes the coupling amount estimation unit 12 and the echo level estimation unit 13, and the echo level integration unit 23 estimates the estimated echo level Z ₁ (ω, obtained by the echo level estimation units 13, 22 _{2 to} 22 _M. n), Z ₂ (ω, n) to Z _M (ω, n) are integrated to generate an estimated echo level Z (ω, n).

As explained in equations (6) and (7), the linear echo component | G (ω) R ₁ (ω) | ² | X (ω, n) | ² in equation (17) is between the speaker and the microphone. Estimated echo level obtained by multiplying the received signal | X (ω, n) | by the estimated value | G '(ω) R' ₁ (ω) |
| Z ₁ (ω, n) | ＝ | G '(ω) R' ₁ (ω) || X (ω, n) | (28)
Can be estimated from

The frequency domain conversion unit 11 for the received signal x (t) converts the received signal x (t) into a frequency domain signal X (ω, n) by the process described with reference to FIG. The coupling amount estimation unit 12 for the received signal x (t) is subjected to the frequency domain conversion for the output X (ω, n) of the frequency domain conversion unit 11 for the received signal x (t) and the main microphone sound reception signal y ₁ (t). from part 15 ₁ of the output Y ₁ (ω, n), the estimated value of the amount of binding between the speaker 31 and the main microphones 32 ₁ | obtained by the following equation (29) | G '(ω ) R' 1 (ω).
| G '(ω) R' ₁ (ω) | ＝ Avg {| Y ₁ (ω, n) | / | X (ω, n) |} (29)

The echo level estimation unit 13 of the received signal receives the output X (ω, n) of the frequency domain transform unit 11 for the received signal x (t) and the output | G ′ (ω) R ′ ₁ (ω ) |, The estimated echo level | Z ₁ (ω) | is obtained by the following equation (29).
| Z ₁ (ω, n) | ＝ Avg {| Y ₁ (ω, n) | / | X (ω, n) |} | X (ω, n) | (30)

The echo level integration unit 23 integrates the estimated echo levels | Z ₁ (ω, n) | to | Z _M (ω, n) | obtained by the echo level estimation units 13, 22 _{2 to} 22 _M to estimate echoes. Output level Z (ω, n). As an integration method, for example, as described in the first embodiment, the maximum value or the average value of the values of each ω may be taken. The configuration and operation of the other parts are the same as in the embodiment of FIG.
Example 3

  FIG. 6 is a block diagram showing an echo canceling apparatus according to a third embodiment of the present invention. The echo canceling apparatus 100 of the present embodiment has a configuration in which an incoming call detection unit 24 and a switch unit 25 are added to the echo estimation unit 20 in the embodiment of FIG.

The reception detection unit 24 observes the level of the output X (ω, n) of the frequency domain conversion unit 11 with respect to the reception signal x (t), and detects a section where the reception is present. In the detection, for example, a fixed threshold value set in advance is compared with the level of the received signal X (ω, n), and when the received signal level is high, it is detected as received. Alternatively, the level ratio between the received signal and the background noise may be detected by comparing with a fixed threshold value. The switch unit 25 is inserted between the echo level integration unit 23 and the echo suppression gain calculation unit 14, and is turned on when an incoming call is detected by the incoming call detection unit 24, and is turned off when no incoming call is detected. The output signal of the level integration unit 23 is turned ON / OFF. When the switch unit 25 is OFF, no reception sound is output from the speaker 31 (thus, no echo is generated). At this time, since the estimated echo level is not given, the echo suppression gain calculation unit 14 gives gain = 1 (or a fixed value) to the gain multiplication unit 16. Accordingly, the received sound signal | Y ₁ (ω) | that does not include an echo is supplied to the inverse frequency domain transform unit 17 as it is.

By these processes, since the switch is OFF in the section where the near-end speaker is uttered and there is no reception, the output of the echo level integration unit 23 is cut off, and the sub-microphone sound reception signal y ₂ (t) Deterioration of the transmitted speech due to the influence of the near-end speaker speech component mixed in ~ y _M (t) is eliminated. Further, when there is an incoming call, the switch is turned on, so that echo can be suppressed with the same performance as the embodiment of FIG. Since the other parts are the same as those in the embodiment of FIG. FIG. 6 shows a case where the reception detection unit 24 detects a reception from the frequency domain reception signal | X (ω) |, but from the level of the reception signal x (t) in the time domain as indicated by a broken line. An incoming call may be detected.

Although the third embodiment has been described as an example in which the reception detection unit 24 and the switch 25 are provided in the embodiment of FIG. 5, the configuration of the reception detection unit 24 and the switch 25 is similarly provided in the embodiment of FIG. May be. In either case shown in FIG. 1 or FIG. 5, the number of sub-microphones may be one. According to the third embodiment, in addition to the effects of the first or second embodiment, it is possible to eliminate the deterioration of the near-end speaker voice in the section where only the near-end speaker voice exists.
Example 4

FIG. 7 is a block diagram showing an echo canceling apparatus according to a fourth embodiment of the present invention.
The echo canceling apparatus 100 of the present embodiment has a configuration in which a main beam former 26 and a sub beam former 27 composed of an M channel fixed filter and an adder are added to the embodiment of FIG. In this embodiment, the M microphones 32 _{1 to} 32 _M are not distinguished from main and sub, and the outputs of the M microphones are given to both the main beam former 26 and the sub beam former 27. The main beamformer 26 has M fixed filter units 26F _{1 to} 26F _M and an adding unit 26A, and coefficients of the fixed filter units 26F _{1 to} 26F _M are set so that the sensitivity becomes higher in the direction of the near-end speaker. . The output of the main beam former 26 is used as a main microphone sound reception signal in each of the embodiments described above. The sub-beamformer 27 also has M fixed filter units 27F _{1 to} 27F _M and an adding unit 27A, and the coefficients of the fixed filter units 27F _{1 to} 27F _M are set so that the sensitivity increases in the direction of the speaker 31. The output of the sub-beamformer 27 is used as a sub-microphone sound reception signal in each of the above embodiments.

The fixed filter units 26F _{1 to} 26F _M of the main beam former 26 filter the M channel microphone sound reception signals y ₁ (t) to y _M (t) with preset filter coefficients, respectively, and the addition unit 26A is fixed. The outputs of the filter units 26F _{1 to} 26F _M are added. The result of addition output from the main beamformer 26 as a main microphone received sound signals provided to the frequency domain transform unit 15 _1. The fixed filter units 27F _{1 to} 27F _{M of the} sub-beamformer 27 also filter the M-channel microphone received signals y ₁ (t) to y _M (t) with preset filter coefficients, respectively. The outputs of the units 27F _{1 to} 27F _M are added. Addition result is output from the sub-beam former 27 as the sub-microphone received sound signals provided to the frequency domain transform unit 15 _2. The configuration and operation of the other parts are the same as in the embodiment of FIG.

In the embodiment of FIG. 1, in the description with reference to FIG. 4, the main microphone 32 ₁ has a high sensitivity direction toward the near-end speaker 40 and a low sensitivity direction toward the speaker 31, and the sub microphones 32 _{2 to} 32 _M have It has been described that the deterioration component of the near-end speaker voice can be reduced by directing the direction with high sensitivity toward the speaker 31 and the direction with low sensitivity toward the near-end speaker 40. In the embodiment of FIG. 7, the main beam former 26 and the sub beam former 27 are used to reduce the deterioration component of the near-end speaker voice. The main beamformer 26 increases sensitivity toward the near-end speaker and decreases sensitivity to the speaker. The sub beam former 27 increases the sensitivity to the speaker 31 and decreases the sensitivity to the near-end speaker. By using the beam formers 26 and 27, it is possible to create a portion having high directivity and a portion having low directivity in an arbitrary direction, and it can be applied to various speaker and microphone arrangements.

ただし、D(ω)は、目標とする周波数応答であり、例えば、振幅値が固定値で、位相が直線位相（時間領域における固定遅延）となっているような応答を設定する。 The design of the fixed filter coefficients of the main beam former 26 and the sub beam former 27 is performed as follows, for example. There are M microphones, and the transfer function from the near-end speaker to the m-th microphone is C _m (ω), and the transfer function from the speaker to the m-th microphone is R _m (ω). However, ω represents a frequency. The coefficient of the m-th channel fixed filter 26F _m of the main beam former 26 is P _m (ω). At this time, what is required of the main beamformer 26 is to pick up the voice of the near-end speaker and suppress the sound of the speaker 31. If these conditions are expressed by equations, equations (31) and (31) (32)

However, D (ω) is a target frequency response, and for example, a response is set such that the amplitude value is a fixed value and the phase is a linear phase (fixed delay in the time domain).

ただし、K(ω)は、目標とする周波数応答であり、例えば、振幅値が固定値で、位相が直線位相（時間領域における固定遅延）となっているような応答を設定する。 Next, the coefficient of the m-th channel fixed filter 27F _m of the sub-beamformer 27 is defined as Q _m (ω). At this time, what is required of the sub-beamformer 27 is to suppress the voice of the near-end speaker and collect the reception speaker sound. If these conditions are expressed by equations, equations (33) and (34) are obtained.

However, K (ω) is a target frequency response, and for example, a response is set such that the amplitude value is a fixed value and the phase is a linear phase (fixed delay in the time domain).

  As described above, if the fixed filter coefficient is set, in the arrangement of an arbitrary microphone and speaker, the main beamformer 26 increases the sensitivity toward the near-end speaker, decreases the sensitivity to the speaker 31, and the subbeamformer 27 It is possible to increase the sensitivity to the speaker 31 and to reduce the sensitivity to the near-end speaker, and to prevent deterioration of the near-end speaker voice.

Also in the embodiment of FIG. 7, as in the embodiment of FIG. 1, the frequency domain converter 11, the coupling amount estimator 12, and the echo level estimator 13 for the received signal x (t) are not used. part 23 also without, may provide an output of the echo level estimator 22 ₂ to the sub-microphone received sound signal as the echo suppression gain computing unit 14. Further, as indicated by a broken line in FIG. 7, the reception detection unit 24 and the switch unit 25 are provided in the same manner as the embodiment of FIG. 6, and the estimated echo level is given to the echo suppression gain calculation unit 14 only during the reception detection period. May be.
As described above, according to the present embodiment, in addition to the effects of the first embodiment or the second embodiment of the present invention, the near-end speaker voice is deteriorated in any speaker and microphone arrangement. It is possible to prevent.

  In each of the embodiments of the echo canceling apparatus according to the present invention described above, the processing for echo suppression for the received signal and the received sound signal inputted as digital signals can be all realized by digital processing. The echo canceling apparatus can be implemented by executing a program for executing the above in a computer.

FIG. 8 shows a basic processing procedure of the echo cancellation method according to the present invention corresponding to the first embodiment of FIG.
Step S1: The main microphone sound reception signal and the sub microphone sound reception signal are converted into frequency domain signals.
Step S2: A level ratio between the frequency domain main microphone received signal and the frequency domain sub microphone received signal is obtained as a coupling amount, and the frequency domain sub microphone received signal is multiplied to obtain an estimated echo level.
Step S3: A ratio between the difference between the level of the frequency domain main microphone sound reception signal and the estimated echo level and the level of the frequency domain main microphone sound reception signal is obtained as a gain.
Step S4: The echo component is suppressed by multiplying the frequency domain main microphone sound reception signal by a gain.
Step S5: The frequency domain main microphone received signal in which the echo component is suppressed is subjected to inverse frequency domain conversion and output as a time domain transmission signal.

Alternatively, in step S1, the main microphone sound reception signal and the plurality of sub microphone sound reception signals are converted into the frequency domain main microphone sound reception signal and the plurality of frequency domain sub microphone sound reception signals, respectively.
In step S2, a level ratio between the frequency domain main microphone sound reception signal and each of the plurality of frequency domain sub microphone sound reception signals is calculated as a coupling amount, and each of the frequency domain sub microphone sound reception signals corresponds to the frequency domain main microphone sound reception signal. A plurality of echo levels may be multiplied to obtain a plurality of echo levels, and the plurality of echo levels may be integrated to generate the estimated echo level.

In the case of the second embodiment shown in FIG. 5, in step S1, the received signal, the main microphone received signal, the at least one sub microphone received signal are converted into a frequency domain received signal and the frequency domain main microphone, respectively. Converted into a sound reception signal and the frequency domain sub-microphone sound reception signal,
In step S2, a level ratio between the level of the frequency domain main microphone received signal and the level of each of the frequency domain received signal and at least one frequency domain received signal is calculated as a coupling amount, and the frequency domain A plurality of echo levels are obtained by multiplying the received signal and the at least one frequency domain sub-microphone received signal by the corresponding coupling amounts, and the estimated echo level is generated by integrating the plurality of echo levels.

  In the case of the third embodiment of FIG. 6, in step S2, the presence / absence of reception is detected based on the level of the reception signal, and the estimated echo level is given to the echo suppression gain calculation unit in the section where reception is detected. The section that is not detected is prohibited.

  In the case of the fourth embodiment of FIG. 7, although not shown in FIG. 8, between the steps S1 and S2, the plurality of microphone sound reception signals are further subjected to first filter processing, respectively. Adding the results and obtaining an addition result as the main microphone sound reception signal; respectively, performing a second filter process on each of the plurality of microphone sound reception signals, adding the results of the second filter process, and adding the addition result to the above And a step of obtaining as a sub-microphone received signal is inserted.

  As described above, according to the first to fourth embodiments of the present invention, it is possible to suppress both the linear acoustic echo and the nonlinear acoustic echo generated due to the distortion characteristic of the speaker, thereby realizing high echo suppression performance. . Further, high-quality sound collection can be realized with little deterioration of the near-end speaker voice. According to the second embodiment of the present invention, the echo suppression performance can be further improved. Further, according to the third embodiment of the present invention, it is possible to eliminate the deterioration of the near-end speaker voice in the section where only the near-end speaker voice exists. Furthermore, according to the fourth embodiment of the present invention, it is possible to prevent the deterioration of the near-end speaker voice in any speaker and microphone arrangement.

1 is a block diagram showing an echo canceller that is a first embodiment of the present invention; FIG. It is a figure which shows the usage of the echo cancellation apparatus of this invention. It is a figure which shows the signal flow of the echo cancellation apparatus of this invention. It is a figure which shows arrangement | positioning of the speaker and microphone for suppressing deterioration of a near-end speaker audio | voice in this invention. It is a block diagram which shows the echo cancellation apparatus which is the 2nd Example of this invention. It is a block diagram which shows the echo cancellation apparatus which is the 3rd Example of this invention. It is a block diagram which shows the echo cancellation apparatus which is the 4th Example of this invention. It is a flowchart which shows the process sequence of the echo cancellation method of this invention. It is a block diagram which shows the conventional echo cancellation apparatus. It is a figure which shows the flow of the signal of the conventional echo cancellation apparatus. It is a figure explaining the conversion procedure of a frequency domain conversion part. It is a figure explaining the conversion procedure of an inverse frequency domain conversion part.

Claims (16)

An echo canceling device that cancels an echo component, which is a signal component output from a speaker and circulated into a microphone, from a microphone received signal,
A frequency domain converting means for converting a main microphone received signal and a sub microphone received signal obtained from a plurality of microphone received signals into a frequency domain main microphone received signal and a frequency domain sub microphone received signal, respectively;
The amount of coupling is estimated based on the level ratio between the frequency domain main microphone received signal and the frequency domain sub microphone received signal, and the estimated echo level is calculated by multiplying the estimated amount of coupling and the frequency domain sub microphone received signal. Echo estimation means for obtaining
An echo suppression gain calculating means for calculating a difference between the level of the frequency domain main microphone received signal and the estimated echo level and a ratio of the level of the frequency domain main microphone received signal as a gain;
A gain multiplier that multiplies the frequency domain main microphone received signal by the gain to suppress echo components in the frequency domain main microphone received signal;
An inverse frequency domain transform unit for performing a reverse frequency domain transform on the multiplication result by the gain multiplier and outputting a time domain transmission signal;
And an echo canceller. The echo canceller according to claim 1,
The frequency domain converting means converts a plurality of sub microphone received signals including the main microphone received signal and the sub microphone received signal into the frequency domain main microphone received signal and the plurality of frequency domain sub microphone received signals, respectively. Includes a plurality of frequency domain transforming units to transform,
The echo estimation unit includes a plurality of coupling amount estimation units that calculate a level ratio between the frequency domain main microphone sound reception signal and each of the plurality of frequency domain sub microphone sound reception signals as a coupling amount, and the plurality of frequency regions. A plurality of echo level estimation units that obtain a plurality of echo levels by multiplying a plurality of corresponding coupling amounts with respect to the sub-microphone received signal, and an echo level that integrates the plurality of echo levels to generate the estimated echo level An echo canceller comprising an integration unit. The echo canceller according to claim 1,
The frequency domain conversion means includes the reception signal, the main microphone reception signal, the at least one sub microphone reception signal, the frequency domain reception signal, the frequency domain main microphone reception signal, and the frequency domain, respectively. It includes a plurality of frequency domain converters that convert sub-microphone sound reception signals,
The echo estimation unit includes a plurality of coupling amount estimation units that calculate, as a coupling amount, a level ratio between the level of the frequency domain main microphone received signal and each level of the frequency domain received signal and the frequency domain received signal. A plurality of echo level estimators for obtaining a plurality of echo levels by multiplying the frequency domain received signal and the at least one frequency domain sub-microphone received signal by a corresponding coupling amount, and the plurality of echo levels. And an echo level integration unit that integrates and generates the estimated echo level. The echo canceller according to claim 3, wherein
The plurality of frequency domain converters include a plurality of frequency domain converters that convert a plurality of sub-microphone sound reception signals including the sub-microphone sound reception signals into a plurality of frequency domain sub-microphone sound reception signals, respectively.
The plurality of coupling amount estimation units includes a plurality of coupling amount estimation units that calculate a level ratio between each of the frequency domain main microphone reception signal and the plurality of frequency domain sub microphone reception signals as a coupling amount. The plurality of echo level estimators includes a plurality of echo level estimators that obtain a plurality of echo levels by multiplying the plurality of frequency domain sub-microphone received signals by a plurality of corresponding coupling amounts, respectively, An echo cancellation apparatus characterized in that an echo level integration unit generates all of the estimated echo levels by integrating all the echo levels.   5. The echo canceller according to claim 1, wherein the echo estimation means includes a reception detection means for detecting presence / absence of reception based on a level of the reception signal, and a section in which reception is detected includes the estimated echo. An echo canceling apparatus comprising: a switching unit that applies a level to the echo suppression gain calculation unit and prohibits a section that is not detected.   The echo canceller according to any one of claims 1 to 5, wherein the plurality of microphones are directional microphones, and among them, the microphone that provides the main microphone sound reception signal has a sensitivity in a transmission sound source direction of the speaker direction. The echo canceling apparatus is characterized in that the microphone that provides the sub-microphone sound reception signal is arranged such that the sensitivity in the speaker direction is higher than the sensitivity in the transmission sound source direction. The echo canceller according to claim 1 or 3, further comprising a main beam former and a sub beam former,
The main beamformer adds a plurality of first filters that respectively filter the plurality of microphone sound reception signals, and outputs of the plurality of first filters, and outputs the addition result as the main microphone sound reception signal. Including an adder,
The sub-beamformer adds a plurality of second filters that respectively filter the plurality of microphone sound reception signals, and outputs of the plurality of second filters, and outputs an addition result as the sub microphone sound reception signal. Including
The coefficients of the plurality of first filters are set to suppress the sound reception signal component in the speaker direction, and the coefficients of the plurality of second filters are set to suppress the sound reception signal component in the transmission sound source direction. An echo canceling device characterized by that. An echo cancellation method for canceling an echo component, which is a signal component output from a speaker and circulated into a microphone, from a microphone reception signal,
(a) a process of converting a main microphone received signal and a sub microphone received signal obtained from a plurality of microphone received signals into a frequency domain main microphone received signal and a frequency domain sub microphone received signal, respectively;
(b) Estimating a coupling amount based on a level ratio between the frequency domain main microphone received signal and the frequency domain sub microphone received signal, and multiplying the estimated coupling amount by the frequency domain sub microphone received signal. The process of obtaining the estimated echo level;
(c) calculating as a gain the ratio between the level of the frequency domain main microphone received signal and the estimated echo level and the level of the frequency domain main microphone received signal;
(d) a process of suppressing an echo component in the frequency domain main microphone received signal by multiplying the frequency domain main microphone received signal by the gain;
(e) a process of outputting a time-domain transmission signal by performing inverse frequency domain transformation on the multiplication result obtained in the process (d),
And an echo canceling method. The echo cancellation method according to claim 8, wherein
In the step (a), a plurality of sub microphone reception signals including the main microphone reception signal and the sub microphone reception signal are converted into the frequency domain main microphone reception signal and the plurality of frequency domain sub microphone reception signals, respectively. Including the process of converting,
The step (b) includes calculating a level ratio between the frequency domain main microphone received signal and each of the plurality of frequency domain sub microphone received signals as a coupling amount, and receiving the plurality of frequency domain sub microphone received signals. Echo cancellation comprising: a step of multiplying a signal by a plurality of corresponding coupling amounts to obtain a plurality of echo levels; and a step of integrating the plurality of echo levels to generate the estimated echo level. Method. The echo cancellation method according to claim 8, wherein
The step (a) includes the reception signal, the main microphone reception signal, the at least one sub-microphone reception signal, the frequency domain reception signal, the frequency domain main microphone reception signal, and the frequency domain, respectively. Including the process of converting to a sub-microphone received signal,
The step (b) includes a step of calculating, as a coupling amount, a level ratio between the level of the frequency domain main microphone received signal and the level of the frequency domain received signal and the frequency domain received signal, and the frequency A process of obtaining a plurality of echo levels by multiplying a region reception signal and at least one frequency domain sub-microphone reception signal by a corresponding coupling amount, and generating the estimated echo level by integrating the plurality of echo levels An echo canceling method comprising the steps of: The echo cancellation method according to claim 10, wherein
In the step (a), a plurality of sub microphone reception signals including the main microphone reception signal and the sub microphone reception signal are converted into the frequency domain main microphone reception signal and the plurality of frequency domain sub microphone reception signals, respectively. Including the process of converting,
The step (b) includes calculating a level ratio between the frequency domain main microphone received signal and each of the plurality of frequency domain sub microphone received signals as a coupling amount, and receiving the plurality of frequency domain sub microphone received signals. Echo cancellation comprising: a step of multiplying a signal by a plurality of corresponding coupling amounts to obtain a plurality of echo levels; and a step of integrating all the echo levels to generate the estimated echo level. Method.   12. The echo canceling method according to claim 8, wherein the step (b) includes a step of detecting presence / absence of reception based on a level of the reception signal, and a section in which reception is detected includes the estimated echo level. The echo canceling method is characterized in that the calculation of the gain according to the step (c) is executed using a step, and the non-detected section includes a switching step for prohibiting the calculation of the gain.   The echo cancellation method according to any one of claims 8 to 12, wherein the plurality of microphones are directional microphones, and among them, the microphone that provides the main microphone sound reception signal has a sensitivity in the direction of the transmission sound source. The echo canceling method is characterized in that the microphone that provides the sub-microphone sound reception signal is arranged such that the sensitivity in the speaker direction is higher than the sensitivity in the transmission sound source direction. The echo cancellation method according to claim 8 or 11, further comprising:
(f) first filtering each of the plurality of microphone sound reception signals, adding the results of the first filter processing, and obtaining an addition result as the main microphone sound reception signal;
(g) a process of performing a second filter process on each of the plurality of microphone sound reception signals, adding the results of the second filter processing, and obtaining an addition result as the sub microphone sound reception signal;
The coefficient of the first filter processing is set to suppress the received signal component in the speaker direction, and the coefficient of the second filter processing is set to suppress the received signal component in the transmission sound source direction. An echo canceling method characterized by being set.   A program for causing a computer to function as the echo canceling device according to any one of claims 1 to 7.   A computer-readable recording medium having recorded thereon a program for causing the computer to function as the echo canceling device according to claim 1.

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