JP2003274481A - Voice processing apparatus, computer program, and recording medium - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a voice processing apparatus capable of quickly eliminating a nonlinear component of an acoustic echo signal and to provide a computer program and a computer-readable recording medium for recording the computer program. <P>SOLUTION: The voice processing apparatus is provided with: a voice signal input terminal 1 for receiving a voice signal from a communication opposite party; a nonlinear inverse system 6 for eliminating in advance nonlinear distortion before a loudspeaker section 3 outputs the received voice signal as a voice; and a linear adaptive filter 5 for estimating a linear component of the acoustic echo signal in a path from the voice signal input terminal 1 to a voice signal output terminal 2, and subtracts the estimated linear component of the acoustic echo signal from the acoustic echo signal generated by a sound reception section 4. <P>COPYRIGHT: (C)2003,JPO

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a voice processing device for reducing an acoustic echo signal generated in two-way voice communication, a computer program for reducing the acoustic echo signal based on an input voice signal, and The present invention relates to a computer-readable recording medium in which the computer program is recorded.

[0002]

2. Description of the Related Art In a device such as a mobile phone with a loudspeaker function and a car phone, a speaker and a microphone can be used as a call destination without using a handset like a video conference system and a hands-free phone. It is possible to talk to the far-end speaker.

In a mobile phone with a voice amplification function, a car phone, etc., a voice signal is transmitted and received through a communication line such as a mobile phone network. At this time, the voice signal transmitted from the call destination is loudened by the speaker and reaches the speaker's ear, and the loud voice is circulated to the microphone, and the voice signal superimposed with the loud sound is transmitted to the call destination. Will be.
That is, a feedback phenomenon occurs in which the voice emitted by the far-end talker to the microphone returns to the ear of the far-end talker, and a so-called acoustic echo occurs.

Since such acoustic echo interferes with a natural call, it needs to be suppressed. Conventionally, in order to suppress acoustic echo, a pseudo acoustic echo signal is generated by estimating an acoustic echo signal in a speaker output propagation path (acoustic echo path) from a speaker to a microphone by an adaptive filter, and the generated pseudo acoustic signal is generated. Acoustic echo canceller devices have been developed that suppress acoustic echoes by subtracting echo signals.

The adaptive filter described above includes FIR (finite
An impulse response) filter is generally used as a linear adaptive filter. It is known that when a linear adaptive filter is used as the adaptive filter, its performance deteriorates when a non-linear factor exists in the acoustic echo path. By the way, since the speaker used in the voice call has non-linearity, there is a problem that the conventional acoustic echo canceller device cannot sufficiently reduce the acoustic echo signal. In particular, a speaker used in a mobile phone with a loudspeaker function has a problem that it is inexpensive and often has a small size, and its non-linearity becomes strong, resulting in significant deterioration of performance.

Therefore, in order to solve the above-mentioned problems,
An acoustic echo canceller device using a non-linear adaptive filter has been developed. FIG. 8 is a block diagram illustrating a conventional acoustic echo canceller device. The acoustic echo canceller device is built in, for example, a mobile phone with a loud sound function, a car phone, or the like. A voice signal transmitted as a digital signal from a call destination is input to a voice signal input terminal 1 through a communication line such as a public telephone line network or a mobile telephone network, and then output as voice to the outside by a loud speaker 3 having a speaker. It Also, the voice of the speaker is input from the sound receiving unit 4 having a microphone and then transmitted from the voice signal output terminal 2 to the call destination through the communication line.

The linear adaptive filter 5a estimates the linear component of the acoustic echo signal generated in the sound receiving section 4 based on the audio signal input from the audio signal input terminal 1 to generate the linear component of the pseudo acoustic echo signal. Similarly, the non-linear adaptive filter 5a estimates the non-linear component of the acoustic echo signal generated in the sound receiving unit 4 and generates the non-linear component of the pseudo acoustic echo signal. The linear component and the non-linear component of the generated pseudo acoustic echo signal are added by the arithmetic processing unit 5c and output.

The pseudo acoustic echo signal generated by adding the linear component and the non-linear component is input to the arithmetic processing unit 7. The arithmetic processing unit 7 performs a process of subtracting the pseudo acoustic echo signal from the acoustic echo signal generated by the voice input unit 4. Therefore, the acoustic echo signal generated by the voice input unit 4 is erased by the arithmetic processing unit 7.

[0009]

However, in the conventional acoustic echo canceller device, a nonlinear adaptive filter called a Volterra filter is used.
Ra filter convergence speed (acoustic echo path identification speed) is generally slow, and since it contains problems such as deterioration of the amount of echo cancellation to be achieved, many practical problems must be overcome. There were challenges left. In particular, the speaker used in the mobile phone with the loud sound function is inexpensive and often small in size, and the sound output from such a speaker has a strong non-linearity. It has been desired to develop an acoustic echo canceller device that can surely and quickly reduce the acoustic echo canceller device.

The present invention has been made in view of the above circumstances, and when the receiving means receives an audio signal, the estimated nonlinear distortion is removed from the audio signal to be output as audio and output from the audio output means. A linear component of the acoustic echo signal generated by the input of the input voice to the voice input means is estimated, and the linear component of the estimated acoustic echo signal is removed from the acoustic echo signal generated by the voice input means. According to the present invention, it is possible to provide a sound processing device, a computer program, and a computer-readable recording medium capable of suppressing a nonlinear component of an acoustic echo signal that may occur and quickly removing the acoustic echo signal. To aim.

[0011]

According to a first aspect of the present invention, there is provided a voice processing device, which receives a voice signal transmitted from a communication device and outputs a voice to the outside based on the voice signal received by the receiving means. A voice output unit for receiving an external voice to generate a voice signal, and a transmitting unit for transmitting the voice signal generated by the voice input unit to the communication device. In a voice processing device configured to reduce an acoustic echo signal generated when the voice output from the output unit is input to the voice input unit, when the voice signal is received by the receiving unit, the voice output unit Means for estimating a non-linear distortion associated with the sound output from the sound signal based on the sound signal, means for estimating a linear component of an acoustic echo signal based on the sound signal, and the estimated non-linear distortion Means for removing from the audio signal to be outputted from the sound output unit, characterized in that it comprises a means for removing the linear component of the estimated acoustic echo signal from the acoustic echo signal generated by said voice input means.

A sound processing apparatus according to a second aspect of the present invention is the sound processing apparatus according to the first aspect of the present invention, further comprising means for calculating a delay time that occurs when estimating the non-linear distortion, and the receiving means for outputting the sound signal. It is characterized in that the linear component of the acoustic echo signal estimated when the calculated delay time has elapsed after reception is removed from the acoustic echo signal generated by the voice input means.

A voice processing device according to a third aspect of the invention is the voice processing device according to the first or second aspect of the invention, wherein
It is characterized in that a non-linear distortion associated with a voice output from the voice output means is estimated by a convolution calculation process with a Volterra series.

A computer program according to a fourth aspect of the present invention causes a computer to estimate a non-linear distortion associated with a voice to be output based on an input voice signal, and causes the computer to estimate the estimated non-linear distortion. A step of causing the computer to output a voice based on the voice signal from which the non-linear distortion is removed, a step of causing the computer to estimate a linear component of the acoustic echo signal based on the input voice signal, Causing the computer to remove the linear component of the estimated acoustic echo signal from the audio signal to be transmitted,
Causing the computer to transmit the audio signal from which the linear component of the acoustic echo signal has been removed.

A computer-readable recording medium according to the fifth aspect of the present invention includes a step of causing a computer to estimate a non-linear distortion associated with a voice to be output based on an input voice signal, and the computer estimates the nonlinear distortion. Removing non-linear distortion from the audio signal; causing a computer to output audio based on the audio signal from which the non-linear distortion is removed; and causing the computer to output the linear sound echo signal based on the input audio signal. The step of estimating the component, the step of causing the computer to remove the estimated linear component of the acoustic echo signal from the voice signal to be transmitted, and the step of causing the computer to transmit the voice signal from which the linear component of the acoustic echo signal has been removed. A computer program is recorded.

According to the first aspect of the invention, when the receiving means receives the audio signal, the nonlinear distortion and the linear component of the acoustic echo signal generated in the audio input means are estimated from the audio signal, and the estimated nonlinear distortion is estimated. Is removed from the voice signal to be output as voice, and the linear component of the acoustic echo signal estimated from the acoustic echo signal generated when the voice output from the voice output means is input to the voice input means is removed. There is. Therefore, for example, by installing a non-linear adaptive filter in consideration of the characteristics of the audio output means such as a speaker in the preceding stage of the audio output means, it is possible to easily remove the non-linear distortion associated with the output audio. . Further, since the nonlinear factor is removed in advance by removing the non-linear distortion from the output voice, the acoustic echo signal that can occur in the voice input means has only a linear component. The linear component of the acoustic echo signal is F
Since it can be easily removed by using a conventionally used linear adaptive filter such as an IR filter, in the present invention, it is not necessary to identify the nonlinear component of the acoustic echo signal, and the acoustic echo signal can be quickly output. Can be reduced. Further, since the non-linear distortion is removed from the output voice, the sound quality of the voice call is improved.

According to the second aspect of the invention, there is provided means for calculating the delay time that occurs when estimating the non-linear distortion.
The linear component of the acoustic echo signal estimated when the calculated delay time has elapsed after receiving the audio signal is removed. Therefore, it is possible to remove only possible acoustic echo signals from the audio signal to be transmitted.

In the third invention, the received voice signal and
The convolution operation with the Volterra series is used to estimate the non-linear distortion associated with the output speech. Therefore, for example, by setting the Volterra series in advance according to the characteristics of the audio output means such as a speaker, it is possible to quickly remove the non-linear distortion.

According to the fourth and fifth aspects of the invention, the non-linear distortion associated with the output voice and the linear component of the acoustic echo signal that can occur are estimated based on the input voice signal, and the estimated voice signal is output from the voice signal to be output. In addition to removing the non-linear distortion, the linear component of the acoustic echo signal is removed from the voice signal to be transmitted. Therefore, for example, by estimating the non-linear distortion in consideration of the characteristics of the audio output means such as a speaker, it is possible to easily remove the non-linear distortion associated with the output sound. In addition, since the nonlinear factor is removed in advance by removing the non-linear distortion from the output voice, the acoustic echo signal that can occur in the voice input means has only a linear component, and the linear component of the acoustic echo signal can be easily converted. It can be removed. According to the present invention, it is not necessary to identify the nonlinear component of the acoustic echo signal, and the acoustic echo signal can be promptly reduced. Further, since the non-linear distortion is removed from the output voice, the sound quality of the voice call is improved.

[0020]

BEST MODE FOR CARRYING OUT THE INVENTION An acoustic echo canceller device embodying a voice processing device of the present invention will be specifically described below with reference to the drawings showing an embodiment thereof. The acoustic echo canceller device is, for example, a mobile phone with a loud sound function,
It is provided in an automobile telephone or the like and has a function of suppressing an acoustic echo generated when a voice output from a voice output unit including a speaker is input to a voice input unit including a microphone.

Embodiment 1. FIG. 1 is a block diagram illustrating an acoustic echo canceller device according to this embodiment. In FIG. 1, reference numeral 1 denotes a voice signal transmitted from a communication terminal device (not shown) such as a telephone or a mobile phone held by a far-end talker, which is a public telephone line network, a communication line such as a mobile telephone network (not shown). Is an audio signal input terminal for receiving via
The voice signal input to the voice signal input terminal 1 is subjected to signal processing by a non-linear inverse system 6 which will be described later, and then output from the loudspeaker 3 as voice to the outside. In addition, the voice uttered by the speaker is input from the sound receiving unit 4, is output as a voice signal from the voice signal output terminal 2, and is transmitted to the communication terminal device of the far-end speaker.

The voice signal transmitted and received through the communication line is preferably a digital signal. However, it is also possible to transmit and receive a voice signal by an analog signal. In that case, an A / D converter for converting the voice signal input to the voice signal input terminal 1 into a digital signal, and the voice signal output terminal 2 It is necessary to include a D / A converter that converts a signal transmitted from the device into an analog signal.

The loudspeaker 3 is provided with a D / A converter, an amplifier and a speaker which are not shown in the figure, and the voice signal processed by the nonlinear inverse system 6 is output as voice. The sound receiving unit 4 includes a microphone, an amplifier, and an A / D converter, which are not shown in the figure, and receives an external sound and generates a digital audio signal.

A delay device 8 and a linear adaptive filter 5 are connected to the path from the audio signal input terminal 1 to the audio signal output terminal 2. The audio signal input to the audio signal input terminal 1 is delayed by the delay unit 8 for a predetermined time (for example, several tens mse
After being delayed by c), it is input to the linear adaptive filter 5. The linear adaptive filter is, for example, an FIR filter, and generates a linear component of the pseudo acoustic echo signal based on the input audio signal.

The linear component of the generated pseudo acoustic echo signal is input to the arithmetic processing unit 7. The arithmetic processing unit 7 performs a process of subtracting the linear component of the pseudo acoustic echo signal from the acoustic echo signal generated by the sound receiving unit 4.

On the other hand, the non-linear inverse system 6 estimates the non-linear distortion associated with the voice output from the loudspeaker based on the input voice signal, and removes the non-linear distortion caused in the loudspeaker 3 in advance. The sound output from the loudspeaker 3 is prevented from being accompanied by non-linear distortion.

In the present invention, as described above, the pseudo acoustic echo signal in which the linear component of the acoustic echo signal generated in the sound receiving unit 4 is estimated is generated and removed, and the non-linear distortion in the transmission path of the voice signal is amplified. By removing from the audio signal output from the unit 3, the nonlinear component of the acoustic echo signal generated in the sound receiving unit 4 is suppressed. Also, the loudspeaker 3
In order to remove the non-linear distortion from the sound output from
The sound quality of the loud sound is improved.

FIG. 2 is a block diagram illustrating the function of the nonlinear inverse system 6. Non-linear inverse system 6 and loudspeaker 3
When there is a non-linear time-invariant causal relationship in the transfer system consisting of and, the input-output relationship of the transfer system is
It can be represented by a series expansion.

[0029]

[Equation 1]

Here, x (n) and y (n) are an input signal and an output signal to the transmission system, respectively, and h ₁
(K ₁ ) is the primary Volterra nucleus and h ₂ (k ₁ , k ₂ ) is 2
It represents the next Volterra nucleus. In this embodiment, a secondary Volterra nucleus is adopted for simplification, and
The nonlinearity of the transfer system is expressed by assuming that the ra kernel has a finite number of taps N.

One object of the present invention is to eliminate the non-linear distortion generated in the loudspeaker 3 (particularly the speaker included in the loudspeaker 3).
It is characterized by being expressed by the convolution of the ra series and the input signal x (n). That is, as shown in FIG. 2, the first-order Volterra filter 31 and the second-order Volter are virtually included in the loudspeaker 3.
This is equivalent to the provision of the ra filter 32, and the arithmetic processing unit 33 adds signals output from the Volterra filters 31 and 32 to virtually generate nonlinear distortion.

On the contrary, the non-linear inverse system 6 estimates the non-linear distortion generated in the loudspeaker 3 and outputs the audio signal obtained by subtracting the non-linear distortion from the input audio signal. Therefore, the audio signal that does not include the nonlinear distortion is output to the arithmetic processing unit 33, and the nonlinear factor in the audio signal can be removed in advance.

The nonlinear inverse system 6 generates the following two types of audio signals. One is to generate an audio signal delayed by a predetermined time by passing through the delay device 61. The audio signal generated here is the second-order Volterra filter 3
By passing through 2, an audio signal with nonlinear distortion delayed by a predetermined time is generated. On the other hand, when an audio signal is input to the model signal generation unit 62, an audio signal simulating a non-linear distortion is generated based on the input audio signal, and the linear inverse filter 63 further causes an audio signal with linear distortion. To generate. The sign of the signal generated here is inverted by the arithmetic processing unit 64 and applied to the delayed audio signal. Then, the audio signal accompanied by the linear distortion and the non-linear distortion is input to the first-order Volterra filter 31, so that the linear distortion is removed first, and then the arithmetic processing unit 33 removes the non-linear distortion.

FIG. 3 is a graph showing the echo cancellation characteristic of the acoustic echo canceller device according to this embodiment. The horizontal axis represents time, and the vertical axis represents the echo cancellation amount. Here, the echo cancellation amount is a certain input signal (for example, colored noise or white noise) y ₀ (k) and an acoustic echo signal y ₁ (k estimated when the input signal y ₀ (k) is input. ) And

[0035]

[Equation 2]

It can be expressed as

In FIG. 3, an echo canceling characteristic of the acoustic echo canceller device according to the present embodiment is shown in a graph 3b. As a comparison target, a simulation using a linear adaptive filter in the acoustic echo canceller device with a linear transmission path is used. Result (graph 3a), a simulation result using a linear adaptive filter in the acoustic echo canceller device with a non-linear transmission path (graph 3d), and the conventional Vo
The result (graph 3c) of the acoustic echo canceller device using the lterra filter is shown.

As shown in FIG. 3, it can be seen that the amount of echo cancellation exceeds 25 [dB] in a relatively short time in the case of the acoustic echo canceller device of this embodiment. When the linear adaptive filter is used in the acoustic echo canceller device with the transmission path being linear (graph 3a), the amount of echo cancellation is most improved, but this is in an ideal state where the non-linearity of the transmission path is not considered. This is because it is, so to speak, a graph showing the performance limit. On the other hand, when a linear adaptive filter is used in the acoustic echo canceller device with a non-linear transmission path (graph 3d), since the non-linearity of the transmission path affects as a disturbance, the echo is compared with the case where the transmission path is assumed to be linear. The erased amount is significantly degraded. Also,
In the conventional acoustic echo canceller device using the Volterra filter, the echo cancellation amount is improved, but the acoustic echo signal calculation speed is very slow. Against these,
It can be seen that the acoustic echo canceller device according to the present embodiment can calculate the acoustic echo signal in a short time, and the echo cancellation amount is improved even compared with the conventional acoustic echo canceller device.

FIG. 4 is a graph comparing the calculation amounts required in the acoustic echo canceller device. The horizontal axis of the graph represents the number of taps (the value of N in Equation 1), and the vertical axis represents the calculation amount required by the acoustic echo canceller device according to the present embodiment divided by the calculation amount required by the conventional Volterra filter. Takes the value

In the acoustic echo canceller device according to this embodiment, when estimating the non-linear distortion generated in the loudspeaker 3, the model signal generator 62 of the non-linear inverse system 6 generates a preset signal. Therefore, regardless of the number N of taps, only a certain amount of calculation is consumed.
Therefore, the overall calculation amount depends on the calculation amount required when the linear component of the acoustic echo signal is estimated by the linear adaptive filter 5, and is a value proportional to the number N of taps. on the other hand,
In the case of the acoustic echo canceller device using the conventional Volterra filter, it is necessary to calculate the quadratic term of the Volterra series, and the calculation amount is proportional to the square of the tap number N.

As a result of the simulation by the inventors,
It is known that the echo cancellation amount can be improved as the number of taps is increased. However, when the total amount of calculation and the delay in the nonlinear inverse system 6 are taken into consideration, the number N of taps is 128. It is practical. When the number of taps N is set to 128, the calculation amount can be reduced to about 1/5 as compared with the conventional one as shown in FIG.

As described above, in the acoustic echo canceller device according to the present embodiment, the amount of calculation is greatly reduced as compared with the conventional acoustic echo canceller device using the Volterra filter, and the echo cancellation amount is improved. . Further, since the non-linear distortion of the audio signal output from the loudspeaker 3 is removed, there is an advantage that the sound quality is improved.

When the acoustic echo canceller device according to this embodiment is incorporated in a small device such as a mobile phone, the linear adaptive filter 5 and the nonlinear inverse system 6 are D
This can be realized by using an SP (digital signal processor), a dedicated LSI, or the like, and a DSP or a dedicated LSI in which the linear adaptive filter 5, the nonlinear inverse system 6, the linear adaptive filter 5, and the delay device 8 are integrated. It is also possible to use.

Further, in the present embodiment, the nonlinear distortion of the audio signal is calculated by using the second-order terms of the Volterra series, but the nonlinear distortion is calculated by using the third-order and higher-order terms. Of course, you can also do it.

Embodiment 2. The above-mentioned acoustic echo canceller device should be applied not only to mobile phones with a loudspeaker function, car phones, etc., but also to systems that use two-way communication of audio signals such as video conference systems and telephone conference systems. Is possible. In this embodiment,
An embodiment applied to a video conference system will be described.

FIG. 5 is a schematic diagram for explaining the video conference system according to this embodiment. In the figure, 100 is a communication device used in a video conference system, and the communication device 100 is a communication network N such as a public telephone line network.
It is connected to the communication device 100 of the communication destination via.

The communication device 100 includes a speaker and a microphone as described later, and the communication network N
It is possible to transmit and receive an audio signal via the communication network N, is equipped with an image pickup device such as a CCD camera and a video camera, and a display device such as a liquid crystal display. It is possible to It should be noted that when transmitting the audio signal and the video data, it is desirable to transmit them in synchronization, but it is not always an essential requirement in the present invention.

In the present embodiment, when a voice signal is received via the communication network N, nonlinear distortion and a linear component of the acoustic echo signal are calculated from the received voice signal by the arithmetic processing of the computer program, and the result is output from the microphone. In addition to removing the non-linear distortion calculated from the audio signal to be output, the linear component of the acoustic echo signal generated in the speaker is also removed.

FIG. 6 is a block diagram showing an internal configuration of the communication device 100 used in the video conference system. The communication device 100 includes a control unit 101 having a CPU, and a ROM 103 and a RAM 10 via a bus 102.
4, the operation unit 105, the display unit 106, the communication unit 107, the loud sound unit 108, the sound receiving unit 109, the imaging unit 110, the auxiliary storage device 111, and other hardware. The control unit 101 controls the hardware according to various control programs such as the program of the present invention, the arithmetic processing program, and the key input processing program stored in the ROM 103. The RAM 104 is configured by SRAM, flash memory, or the like, and temporarily stores data generated when various control programs stored in the ROM 103 are executed, video data transmitted and received by the communication unit 107, and the like.

The operation section 105 is provided with hardware keys or software keys such as ten keys and function keys necessary for operating the communication device 100. The display unit 106 includes a display device such as a liquid crystal display, and displays the video data and the like received by the communication unit 107.

The communication section 107 is provided with a line terminating device such as a modem, and receives the audio signals input to the loudspeaker section 108 and the sound receiving section 109, the video data input to the image pickup section 110, and the like from the communication network N. It controls when sending and receiving through.

The loudspeaker 108 has a D / A (not shown).
The control unit 101 includes a converter, an amplifier, and a speaker.
After the signal processing is performed by, the voice signal is output as voice. The sound receiving unit 4 includes a microphone, an amplifier, and an A / D converter, which are not shown, and receives an external sound and generates a sound signal as a digital signal.

The image pickup unit 110 has an image pickup device such as a CCD camera or a video camera, picks up an image of a speaker, obtains video data, and transmits it to the communication device 100 of the communication destination through the communication network N. .

The auxiliary storage section 111 is composed of a CD-ROM drive or the like for reading the computer program and data from a recording medium 112 such as a CD-ROM in which the computer program and data of the present invention are recorded. The read computer program and data are , ROM 103. The computer program of the present invention is
The form provided by the recording medium 112 is not necessarily required, and the form stored in the ROM 103 in advance may be used.

FIG. 7 is a flow chart showing a processing procedure of a voice signal by the communication device 100. Control unit 101
First, the communication unit 107 determines whether or not an audio signal is received (step S1). When the voice signal is not received (S1: NO), it waits until the voice signal is received.

When a voice signal is received (S1: YE
S), based on the received voice signal, a non-linear distortion associated with the voice output by the loudspeaker 108 is calculated (step S2). When calculating the non-linear distortion, the Volterra series 2 preset in consideration of the characteristics of the loudspeaker 108 is used.
It is calculated by the learning identification method (NLMS method) using the following terms. Further, the linear component of the acoustic echo signal is calculated based on the received voice signal (step S3). When calculating the linear component of the acoustic echo signal, in addition to the learning identification method described above,
Conventional methods such as the LMS algorithm and RLS algorithm can be used.

Next, the control unit 101 removes the non-linear distortion calculated in step S2 from the received voice signal (step S4), and outputs the non-linear distortion-removed voice signal from the loudspeaker unit 108 (step). S
5).

Next, the control unit 101 determines whether or not a predetermined time has elapsed after receiving the audio signal in step S1 (step S6). When it is determined that the predetermined time has not elapsed (S6: NO), the process stands by until the predetermined time elapses. The predetermined time corresponds to the delay time that occurs when the nonlinear distortion is calculated in step S2, and when calculating the linear component of the acoustic echo signal, the time calculated based on the input audio signal. Is.

When it is determined that the predetermined time has passed (S
6: YES), the linear component of the acoustic echo signal is removed from the sound signal received by the sound receiving unit 109 (step S).
7). Then, after removing the linear component of the acoustic echo signal, the audio signal is transmitted from the communication unit 107 (step S
8).

In the present embodiment, the non-linear distortion of the audio signal and the linear component of the acoustic echo signal are removed by the processing of the computer program. It can also be realized by using an LSI.

[0061]

As described above in detail, according to the first aspect of the present invention, when the receiving means receives the audio signal, the audio signal has a non-linear distortion and the linear component of the acoustic echo signal generated in the audio input means. Is estimated, and the estimated nonlinear distortion is removed from the voice signal to be output as voice, and the acoustic echo signal estimated from the acoustic echo signal generated when the voice output from the voice output means is input to the voice input means. The linear component of is removed. Therefore, for example, by installing a non-linear adaptive filter in consideration of the characteristics of the audio output means such as a speaker in the preceding stage of the audio output means, it is possible to easily remove the non-linear distortion associated with the output audio. . Further, since the nonlinear factor is removed in advance by removing the non-linear distortion from the output voice, the acoustic echo signal that can occur in the voice input means has only a linear component. Since the linear component of the acoustic echo signal can be easily removed by using a conventionally used linear adaptive filter such as an FIR filter, the present invention needs to identify the nonlinear component of the acoustic echo signal. Therefore, the acoustic echo signal can be reduced promptly. Further, since the non-linear distortion is removed from the output voice, the sound quality of the voice call is improved.

According to the second aspect of the invention, means for calculating the delay time generated when estimating the non-linear distortion is provided, and the acoustic echo estimated when the calculated delay time elapses after receiving the audio signal. The linear component of the signal is removed. Therefore, it is possible to remove only possible acoustic echo signals from the audio signal to be transmitted.

In the case of the third aspect of the invention, the non-linear distortion associated with the output voice is estimated by the convolution operation processing of the received voice signal and the Volterra series. Therefore, for example, by presetting the Volterra series according to the characteristics of the audio output means such as a speaker,
It is possible to quickly remove the non-linear distortion.

In the case of the fourth and fifth inventions, the nonlinear distortion associated with the output voice and the linear component of the acoustic echo signal which can occur are estimated based on the input voice signal, and the nonlinear component is estimated from the voice signal to be output. Is removed, and the linear component of the acoustic echo signal is removed from the voice signal to be transmitted. Therefore, for example, by estimating the non-linear distortion in consideration of the characteristics of the audio output means such as a speaker, it is possible to easily remove the non-linear distortion associated with the output sound. In addition, since the nonlinear factor is removed in advance by removing the non-linear distortion from the output voice, the acoustic echo signal that can occur in the voice input means has only a linear component, and the linear component of the acoustic echo signal can be easily converted. It can be removed.
According to the present invention, it is not necessary to identify the nonlinear component of the acoustic echo signal, and the acoustic echo signal can be promptly reduced. Further, since the non-linear distortion is removed from the output voice, the present invention has an excellent effect such that the sound quality of the voice call is improved.

[Brief description of drawings]

FIG. 1 is a block diagram illustrating an acoustic echo canceller device according to this embodiment.

FIG. 2 is a block diagram illustrating the function of a nonlinear inverse system.

FIG. 3 is a graph showing echo cancellation characteristics of the acoustic echo canceller device according to the present embodiment.

FIG. 4 is a graph comparing the calculation amounts required in the acoustic echo canceller device.

FIG. 5 is a schematic diagram illustrating a video conference system according to the present embodiment.

FIG. 6 is a block diagram showing an internal configuration of a communication device used in the video conference system.

FIG. 7 is a flowchart showing a processing procedure of an audio signal by the communication device.

FIG. 8 is a block diagram illustrating a conventional acoustic echo canceller device.

[Explanation of symbols]

1 Audio signal input terminal 2 Audio signal output terminal 3 loudspeaker 4 sound receiving section 5 Linear adaptive filter 6 Non-linear inverse system 7 Arithmetic processing section 8 delay device

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Claims (5)

[Claims] 1. A receiving means for receiving a voice signal transmitted from a communication device, a voice output means for outputting a voice to the outside based on the voice signal received by the receiving means, and a means for receiving an external voice. A voice input means for generating a voice signal and a transmitting means for transmitting the voice signal generated by the voice input means to the communication device are provided, and the voice output from the voice output means is input to the voice input means. In a voice processing device configured to reduce an acoustic echo signal caused by the above, when a voice signal is received by the receiving unit, a non-linear distortion associated with a voice output from the voice output unit is generated in the voice signal. Based on the audio signal, a means for estimating a linear component of an acoustic echo signal based on the audio signal, and an estimated nonlinear distortion from the audio signal to be output from the audio output means. Speech processing apparatus characterized by comprising: means for to, and means for removing a linear component of the estimated acoustic echo signal from the acoustic echo signal generated by said voice input means. 2. An acoustic echo signal estimated when a delay time calculated after the audio signal is received by the receiving means is provided with a means for calculating a delay time generated when the nonlinear distortion is estimated. 2. The audio processing apparatus according to claim 1, wherein the linear component of is removed from the acoustic echo signal generated by the audio input means. 3. A non-linear distortion associated with a voice output from the voice output means is estimated by a convolution operation process of a voice signal and a Volterra series. 2. The voice processing device according to 2. 4. A step of causing a computer to estimate a non-linear distortion associated with a voice to be output based on an input voice signal; a step of causing the computer to remove the estimated non-linear distortion from the voice signal; To output a voice based on the voice signal from which the non-linear distortion is removed, a step of causing the computer to estimate a linear component of the acoustic echo signal based on the input voice signal, and a computer to estimate the estimated acoustic echo. A computer program comprising: a step of removing a linear component of a signal from an audio signal to be transmitted; and a step of causing a computer to transmit an audio signal from which a linear component of an acoustic echo signal has been removed. 5. A step of causing a computer to estimate a non-linear distortion associated with a voice to be output based on an input voice signal, a step of causing the computer to remove the estimated non-linear distortion from the voice signal, To output a voice based on the voice signal from which the non-linear distortion is removed, a step of causing the computer to estimate a linear component of the acoustic echo signal based on the input voice signal, and a computer to estimate the estimated acoustic echo. A computer program recorded with a computer program having a step of removing a linear component of a signal from an audio signal to be transmitted and a step of causing a computer to transmit an audio signal from which a linear component of an acoustic echo signal is transmitted. Readable recording medium.