Telephonic communications – Subscriber line or transmission line interface – Network interface device
Patent
1996-11-13
1998-12-08
Hunter, Daniel S.
Telephonic communications
Subscriber line or transmission line interface
Network interface device
379406, 379411, 381 711, H04M 900
Patent
active
058481516
DESCRIPTION:
BRIEF SUMMARY
BACKGROUND OF THE INVENTION
The present invention relates to acoustical echo cancellers for telephone terminals used under conditions in which the signal output to a telephone line is affected by an acoustical echo, often together with noise due to the surroundings. A major application lies in installations that may include portable hands-free terminals and in teleconferencing installations.
Adaptive transversal filters operating in the time domain are often used in acoustical echo cancellers. Implementation thereof encounters numerous difficulties. The number of coefficients required for modelling the impulse response of an acoustical echo path is very high. Speech signals have a high degree of self-correlation and they are non-stationary.
Finally, the signal received by the input sensor (usually a microphone) includes not only the acoustical echo, but also room noise and near-end speaker speech, in the case of simultaneous speech.
To take account of the length of the impulse response and the high degree of correlation, proposals have already been made to use filters implementing an algorithm that operates in the frequency domain. However, problems remain with resistance to room noise and with taking account of the situation when both the near-end speaker and the far-end speaker are speaking simultaneously.
Attempts have been made to improve the characteristics of adaptive transversal filters operating in the time domain or in the frequency domain. In the time domain, proposals have been made in particular to apply the stochastic gradient algorithm (also known as the least mean square (LMS) algorithm) to blocks of samples rather than sample by sample. The gradient is then estimated block by block and provides an indication concerning the correlation between the input signal and the error signal on the corresponding block. Adaptive echo cancellers have also been proposed operating in the frequency domain, based on fast convolution techniques of the overlap-and-save (OLS) type or of the overlap-and-add (OLA) type. FIG. 1 is a block diagram of an adaptive filter usable in an echo canceller of this kind. The incoming digital signal x(n), where n is the order of the sample, is applied to a loudspeaker 12 after analog-to-digital conversion ADC and amplification by an amplifier 10. The acoustical echo arrives via a path 13 to the detector 14 which is generally constituted by a microphone. The microphone also receives room noise and, at certain moments, a speech signal from the near speaker. The signal amplified at 16 is put into digital form z(n) by an analog-to-digital converter 22.
The near terminal echo canceller is interposed between the receive line LR receiving the incoming signal and the send line LE supplying the outgoing signal. Because the filter operates on blocks, it includes in the send line LE an input serial-to-parallel converter 23 and an output parallel-to-serial converter 24, with an adder 26 being interposed therebetween which receives the input sample blocks on an additive input, and on a subtractive input receives the correction signal blocks as generated by an adaptive filter 28.
The echo canceller shown in FIG. 1 operates in the frequency domain and it uses a partition-and-overlap procedure. This amounts to computing a Fourier transform (or more generally a transform going from the time domain to the frequency domain), of a size equal to twice the length of the estimated impulse response N. The filter 28 receives the incoming signal x(n) via a serial-to-parallel converter 32 enabling blocks to be built up, a circuit 34 for partitioning into blocks with overlap, and a circuit 36 for computing the discrete Fourier transform (DFT). Symmetrically, the output from the filter 28 is applied to a circuit 38 for computing the inverse Fourier transform (DFT.sup.-1), and thence via an overlap circuit 40 for recovering the last block, to the adder 26.
The means for adapting the coefficients of the frequency filter 28 are of known general structure. They comprise a multiplier 42 which receives the out
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Boudy Jerome
Capman Fran.cedilla.ois
Hunter Daniel S.
Matra Communications
Saint-Surin Jacques
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