JPH0766757A - Method for canceling echo - Google Patents

Abstract

PURPOSE:To surely detect double talking and to improve an echo elimination rate. CONSTITUTION:A sound signal encoded by a CELP system is decoded by a decoder 13, the decoded signal is supplied to a pseudo echo line 24 and an impulse response estimating part 25, D/A converted and then supplied to an echo line 23. A signal from the echo line 23 side is A/D converted, a signal outputted from the pseudo echo line 24 is subtracted from the A/D converted signal by an erasing circuit 18 and the result is supplied to a sound encoder 19 and also to the estimating part 25. From a spectrum envelope decoder 47 in the decoder 13, a spectrum envelope and pitch period power are extracted, which are supplied to an auxiliary information comparing part 55. The output of the A/D converter 17 is supplied to an LPC analyzing part 56 and its spectrum envelope and pitch period power are detected and supplied to the comparing part 55. When a distance between both the spectrum envelopes is long, the comparing part 55 judges the existence of double talk and aborts impulse response estimation, and when the distance is short and both pitch periods are almost equal, an estimation speed is accelerated by increasing the correction step size of impulse response estimation.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an echo canceling method for canceling echo signals which are a cause of howling and an auditory obstacle in a loudspeaker telephone conference communication system, a two-wire to four-wire conversion system and the like.

[0002]

2. Description of the Related Art FIG. 4A shows a loudspeaker type communication terminal device equipped with a high-efficiency speech coder / decoder. The signal from the transmission line received through the input terminal 11 is decoded by the transmission line decoder 12 into a baseband signal, and the baseband signal is converted into a coded voice signal by a voice decoder 13 into, for example, a telephone band voice signal. It is decoded and further converted into an analog signal by the D / A converter 14. This analog audio signal is sent to the speaker 1
5 and is emitted as an acoustic signal. On the other hand, the voice signal received by the microphone 16 is converted into a digital signal by the A / D converter 17, the echo signal is eliminated by the erasing circuit 18, and the voice signal is supplied to the voice coder 19 for high efficiency voice encoding. The encoded voice signal is encoded by the transmission line encoder 21 into a signal on the transmission line and transmitted from the output terminal 22 to the transmission line. In order to prevent the acoustic signal emitted from the speaker 15 from being captured by the microphone 16 and transmitted as an echo signal, the pseudo echo path 24 that imitates the echo path 23 connecting the speaker 15 and the microphone 16 is a speaker. The signal to the speaker 15 is branched and supplied to the pseudo echo path 24, the output passing through the pseudo echo path 24 is supplied to the cancellation circuit 18, and is subtracted from the signal from the microphone 16, that is, the echo signal is canceled. To be done. The input signal of the speaker 15 and the output signal of the erasing circuit 18 are input to the impulse response estimation unit 25, the impulse response of the echo path 23 is estimated, and the estimated impulse response characteristic is set in the pseudo echo path 24. Echo path 24
The impulse response is convolved with the signal input to the.

Similarly, in a 4-wire / 2-wire conversion system, FIG.
4A, the output side of the D / A converter 14 and the input side of the A / D converter 17 are connected to the 4-wire side terminal of the hybrid transformer 26, as indicated by the same reference numerals. The 2-wire type transmission line 27 is connected to the 2-wire side terminal of the hybrid transformer 26. There is an echo path 28 in which the output signal of the D / A converter 14 leaks from the hybrid transformer 26 and reaches the A / D converter 17 side, and the echo signal passing through this echo path 28 is canceled by the canceling circuit 18 in the case of FIG. 6A. Similarly, it is canceled.

Further, as shown in FIG. 5, in a base station 29 for mobile radio communication, a digital voice signal from an analog network 31 is encoded by a voice encoder 19 and further encoded by a transmission line encoder 21. In the mobile terminal device 32, the signal of the base station 29 is transmitted to the mobile terminal device 32 via a wireless line, and the signal of the base station 29 is converted into a baseband signal by the transmission path decoder 33, and further decoded into a voice signal by the voice decoder 34
The audio signal is converted into an analog signal by the D / A converter 14 and supplied to the speaker 15. The audio signal from the microphone 16 is converted into a digital signal by the A / D converter 17,
The voice encoder 35 performs high-efficiency encoding, and the encoded output is converted into a code signal on the transmission line by the transmission line encoder 36 and transmitted to the base station 29 via a wireless line. In the base station 29, the received signal is decoded into a baseband signal in the transmission line decoder 12, and the baseband signal is decoded into a digital voice signal in the voice decoder 13 to obtain the analog network 31.
Sent to. Also in this case, the echo path 23 from the speaker 15 to the microphone 16 is formed, and the cancellation of the echo signal through the echo path 23 is performed by the voice encoder 19 of the base station 29.
Is performed by the pseudo echo path 24, the erasing circuit 18, and the impulse response estimation unit 25 provided between the input side of the ∘ and the output side of the speech decoder 13.

The speech coder and speech decoder shown in FIGS. 4A, 4B, and 5 are those for encoding and decoding speech signals with high efficiency using linear prediction, and for example, CELP (Code).
Excited Linear Predictio
n: code-excited linear prediction) coding method is used. To briefly describe this, the input audio signal is L as shown in FIG. 6A.
The PC analysis unit 41 performs LPC analysis to obtain a spectrum envelope parameter for each block, and this parameter is set as a filter coefficient in the linear prediction synthesis filter 42. The excitation signal selected from the excitation source 43 is given a gain in the gain section 44 and is supplied to the linear prediction synthesis filter 42 as an excitation signal. The distortion evaluation unit 45 performs selection of the excitation signal of the excitation source 43 and gain control given to the gain unit 44 so that the distortion of the synthesized signal synthesized by the synthesis filter 42 with respect to the input speech signal is minimized. A code indicating an excitation signal (vector) in which an audio signal is selected in block units, a code indicating a set gain, and a spectrum envelope parameter are output as a coded signal.

As shown in FIG. 6B, the decoder for decoding the coded signal takes out the spectrum envelope parameter in the spectrum envelope decoder 47, sets it as the filter coefficient in the linear prediction synthesis filter 48, and sets the excitation source. The excitation signal is selectively decoded by the decoder 49, and the excitation signal is provided with the gain decoded by the gain unit 51 and is input to the linear prediction synthesis filter 48 as the excitation signal.
The audio signal is restored and output from 8.

Although the conditions required for echo cancellation differ between the acoustic echo (FIG. 4A) and the line echo (FIG. 4B), since the principle of echo cancellation is common, the acoustic echo cancellation method will be described below. The impulse response characteristic of the pseudo echo path 24 is sequentially updated based on the residual echo of the output of the canceling circuit 18. This update is performed when the other party is not speaking, and in other cases, the impulse response characteristic is updated. Need to be frozen. Therefore, the speaker 1 is conventionally used.
The input of 5 and the output of the microphone 16 are monitored to determine the state in which the other party is speaking, that is, the state in which the other party's voice is input in addition to the echo signal from the echo path 23. It was

[0008]

However, if the volume of the speaker of the other party or the positions of the speaker 15 and the microphone 16 change, it is not possible to correctly determine the state in which the other party is talking (double talk state). It was For this reason,
The impulse response is not correctly estimated, and the pseudo echo path 2
The characteristic of 3 may be disturbed. In order to prevent such a situation, the estimation calculation of the impulse response of the echo path 23 cannot be rapidly converged in the past.

[0009]

According to the present invention, at least one of a spectrum envelope parameter and a pitch period parameter is extracted by performing a linear predictive analysis on both the signal to the echo path and the signal from the echo path. , The difference between these two extracted parameters is detected, and the estimated speed of the impulse response is adaptively controlled according to the difference. That is, when the difference is sufficiently large, the impulse response estimation process is stopped, and when the difference is sufficiently small, the estimation process is performed. At this time, it is advisable to also monitor the respective sound volumes of the signal to the echo path and the signal from the echo path, and also to judge whether or not the state is the double-talk state based on this.

When a signal to the echo path and a signal from the echo path are coded and decoded with high efficiency using linear prediction, a spectrum envelope parameter or pitch period used for the coding and decoding is used. The parameters are used for the double talk detection.

[0011]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS FIG. 1 shows an embodiment of the invention of claim 4, and the same reference numerals are given to the portions corresponding to those in FIGS. In this embodiment, the so-called auxiliary information decoded from the spectrum envelope decoder 47 in the speech decoder 13, that is, the spectrum envelope parameter, and the decoded pitch period parameter and power (volume) parameter are compared to the auxiliary information comparison unit 5
Supply to 5. Further, the output signal of the A / D converter 17 is branched and supplied to the LPC analysis unit 56 to obtain the spectrum envelope parameter, the pitch period parameter and the power parameter, and these parameters are input to the auxiliary information comparison unit 55. The auxiliary information comparison unit 55 detects a difference between corresponding ones of the two inputs, and adaptively controls the estimated speed of the impulse response estimation unit 25 according to the difference.

For example, if the difference (distance) between the two corresponding spectral envelopes is detected, and if this is large, it is considered that double talk or large disturbance has occurred, and the impulse response estimation process is stopped. If the difference in spectral envelope is small and both pitch periods match, the reverberation signal will surely wrap around, it will not be in the state of double talk, it will be judged that the ambient noise is also small, and the estimated speed of the impulse response will be increased, that is, Increase the correction step size to speed up convergence. The estimation speed of the impulse response is slowed in the situation where the difference in the spectral envelope and the difference in the pitch period are intermediate such that there is a large amount of ambient noise or the possibility of double talk.

In contrast to the signal to the echo path 23, the echo signal from the echo path 23 has a delay that can be calculated by the system and the echo path 2
It is delayed by 3 delays. Further, the echo signal from the echo path 23 is subjected to the deformation of the frequency response estimated to the impulse response of the echo path 23 with respect to the signal to the echo path 23. Therefore, as shown in FIG.
The auxiliary information decoded from is supplied to the auxiliary information comparison unit 55 through the correction unit 57, and the correction unit 57
The delay is delayed by the delay amount, and a deformation corresponding to the deformation of the frequency response is given. The delay amount and the frequency response are modified by the impulse response estimation unit 25. Only the delay correction may be performed, and similarly, only the frequency response deformation of the spectrum envelope may be corrected. The correction for the frequency response deformation may be performed on the spectrum envelope from the LPC analysis unit 56. In this case, the correction is performed so as to remove the frequency response deformation in the echo path 23.

An example in which the present invention is applied to the echo canceling system shown in FIG. 5 is shown in FIG. 3 in which parts corresponding to those in FIGS. In this embodiment, the spectrum envelope parameter, the pitch period parameter, and the power parameter from the LPC analysis unit 41 in the encoder 19 are supplied to the auxiliary information comparison unit 55, and the decoded speech signal of the speech decoder 13 is output. The spectrum envelope parameter, the pitch period parameter, and the power parameter are supplied to the LPC analysis unit 56, and are supplied to the auxiliary information comparison unit 55. Others are the same as the above description. The encoder 19 has a large number of voice encodings and decodings and a large amount of quantization noise.
Is decoded by the local decoder 58 as shown by the dotted line, and its spectrum envelope parameter, pitch period parameter,
The power parameter may be supplied to the auxiliary information comparison unit 55.

Further, the decoded excitation signal may be supplied instead of the decoded speech signal as shown by the dotted line in FIG. The frequency characteristic of the excitation signal is almost flat, so that the impulse response is estimated in a relatively short time. In this case, the excitation signal may be passed through the bandwidth expansion synthesis filter controlled by the decoded spectrum envelope parameter and supplied to the impulse response estimation unit 25 as a gently whitened signal. The bandwidth expansion coefficient of the bandwidth expansion synthesis filter is, for example, about 0.5. Supplying this excitation signal to the impulse response estimation unit 25 can be applied to other embodiments. In the above description, the present invention is applied to the case where the audio coding and the decoding are performed on the signal to the echo path and the signal from the echo path, but even when such coding and decoding are not performed,
The present invention can be applied by extracting at least one parameter of the spectrum envelope and the pitch period for the signal to the echo path and the signal from the echo path.

[0016]

As described above, according to the present invention, double talk can be accurately detected, impulse response estimation is stopped in that state, and there is no fear of disturbing the characteristics of the pseudo echo path. You can also know the state that seems to be double talk, and in this state slow the estimation degree of the impulse response,
The estimation speed of the impulse response can be increased while the echo signal can be detected accurately.

Particularly when used in combination with speech coding and decoding based on linear prediction, double-talk detection processing uses the parameters of the spectrum envelope and pitch period used for speech coding and decoding as they are. It is possible,
The increase in the amount of calculation is small.

[Brief description of drawings]

FIG. 1 is a block diagram showing an embodiment of the present invention.

FIG. 2 is a block diagram showing another embodiment of the present invention.

FIG. 3 is a block diagram showing still another embodiment of the present invention.

4A is a block diagram showing a conventional acoustic echo canceller in a loudspeaker type communication terminal, and FIG. 4B is a block diagram showing a conventional line echo canceller.

FIG. 5 is a block diagram showing a conventional configuration for canceling remote echo.

FIG. 6A is a block diagram showing an example of a speech encoder 19.
B is a block diagram showing an example of the speech decoder 13.

Front page continuation (72) Inventor Shoji Shimada 1-1-6 Uchisaiwaicho, Chiyoda-ku, Tokyo Nihon Telegraph and Telephone Corporation

Claims (4)

[Claims] 1. A signal to the echo path is supplied to the pseudo echo path to convolve the impulse response, the output of the pseudo echo path is subtracted from the signal from the echo path, and the subtracted output and the echo path to the echo path. In the echo cancellation method of estimating the impulse response of the echopath from the signal and controlling the characteristics of the pseudo echopath with the estimated impulse response, in both the signal to the echopath and the signal from the echopath The linear envelope analysis to the spectral envelope parameter,
An echo canceling method, which comprises extracting at least one of pitch period parameters, detecting a difference between the two extraction parameters, and adaptively controlling an estimated speed of the impulse response according to the difference. 2. A difference between the spectrum envelope parameters is obtained as a difference between the two extraction parameters, and one of the spectrum envelope parameters is corrected according to the characteristic of the pseudo echo path, and then the difference is obtained. The echo canceling method according to claim 1. 3. The echo canceling method according to claim 1, wherein a delay of the impulse response on the pseudo echo path is calculated, and the difference between the extraction parameters is detected in consideration of the delay. 4. The transmitted and received voice signals for the echo path are coded and decoded with high efficiency using linear prediction, and the spectrum envelope parameter or pitch period parameter used in the coding or decoding is calculated. The echo canceling method according to any one of claims 1 to 3, wherein the echo canceling method is used as the spectrum envelope parameter or the pitch period.