Telephonic communications – Echo cancellation or suppression
Reexamination Certificate
1998-07-29
2001-06-19
Korzuch, William R. (Department: 2641)
Telephonic communications
Echo cancellation or suppression
Reexamination Certificate
active
06249581
ABSTRACT:
FIELD OF THE INVENTION
The present invention relates generally to the field of acoustic echo cancellation in telecommunications, and particularly to a pseudo spectrum-based acoustic echo canceller which adaptively cancels echoes arising in hands-free audio and video teleconferencing and related systems without requiring a state machine or training.
BACKGROUND OF THE INVENTION
Acoustic echo cancellers and their applications in the field of telecommunication are well known to those skilled in the art. Many such cancellers and related technologies have been described in various publications including the following patent documents:
U.S. Pat. No. 5,548,642
U.S. Pat. No. 5,530,724
U.S. Pat. No. 5,506,901
U.S. Pat. No. 5,428,562
U.S. Pat. No. 5,406,583
U.S. Pat. No. 5,394,392
U.S. Pat. No. 5,384,806
U.S. Pat. No. 5,329,586
U.S. Pat. No. 5,206,854
U.S. Pat. No. 5,163,044
U.S. Pat. No. 5,146,494
U.S. Pat. No. 5,016,271
U.S. Pat. No. 5,001,701
U.S. Pat. No. 4,918,685
U.S. Pat. No. 4,817,081
U.S. Pat. No. 4,464,545
A typical acoustic echo canceller currently available uses what-is-known-as an adaptive filter which employs a well-known algorithm such as the algorithm known as the Least-Mean-Square algorithm, or LMS. This algorithm continuously adapts to changes in the placement of both the speaker and microphone and to changes in loudspeaker volume. For these cancellers, a state machine is needed to automatically determine each of the four states, i.e., receiving, transmitting, double-talk, and idle. In addition, in order to cancel the echoes, these cancellers much be trained, that is, they must “learn” the loudspeaker-to-microphone acoustic response function for the room it is servicing. Also, the acoustic compensation length is determined by the length of the filter that is determined by the host resource availability.
OBJECT OF THE INVENTION
It is an object of the present invention to provide an acoustic echo canceller which adaptively cancels echo arising in hands-free audio and video teleconferencing systems and other related systems where echo cancellation is required.
It is an another object of the present invention to provide an acoustic echo canceller which provides high-quality and low cost full duplex speech communication typical of dedicated video conferencing systems.
It is yet another object of the present invention to provide an acoustic echo canceller which does not require a state machine.
It is still yet another object of the present invention to provide an acoustic echo canceller which does not require training.
It is still yet another object of the present invention to provide an acoustic echo canceller which continuously adapts to changes in microphone and loudspeaker placement, loudspeaker volume setting, and the movement of people.
It is still yet another object of the present invention to provide an acoustic echo canceller which is independent of any standard.
It is still yet another object of the present invention to provide an acoustic echo canceller which can be connected directly to a PC soundcard and an ordinary telephone set.
SUMMARY OF THE INVENTION
A microphone array is used together with a block adaptive algorithm to effectively suppress acoustic echo arising in hands free voice communication. A the same time, the system is also capable of suppressing environmental noise.
The present echo canceller utilizes the principle that the spectrum pattern of human speech does not change much in the short run. The present echo canceller takes 256 overlap 128 samples in 16 ms intervals, or sample blocks. The power spectrum taken at time 0 and at any time within the 16 ms interval are essentially the same. This is true even though the waveform of the speech may change over time even in the short run. The echoes are simply a delayed form of a speech signal. Therefore, in following the principle described above, the spectrum of the speech signal and the spectrum of the echo taking are substantially the same.
The inputs to the present echo canceller are x(t) and y(t), y(t) representing the incoming speech signal from a far-end speaker and x(t) representing the combination of speech signal from a near-end speaker and the echo. The well-known normalized cross-correlation estimation between x(t) and y(t) is performed to determine the level of correlation between x(t) and y(t) which is quantitatively represented by the correlation coefficient C, a value of 1 for C being perfect correlation.
When the far-end speaker is speaking and the near-end speaker is not speaking, x(t) comprises of only the echo portion which is essentially a delayed form of y(t). In that case, there is almost a perfect correlation between x(t) and y(t) and the C value is near 1. When the near-end speaker is speaking and the far-end speaker is not speaking, the x(t) comprises only of the signal and the C value is near 0. When both the near-end and the far-end speakers are speaking simultaneously, the C value may be between 0 and 1, but typically near to 0 since the two speech signals will not be highly correlated. And of course, silence would result in a near 0 also, since respective noises will not be highly correlated. Certain decisions are based on whether the C value exceeds certain thresholds.
Since the echo is essentially a delayed y(t), the amount of delay is estimated by measuring the time shift required to produce the maximum C value. Once the delay is determined, the two channels of inputs are aligned by time-shifting x(t) to match y(t). The amplitude of the x(t) and y(t) is then normalized by first determining a certain gain factor, and then multiplying y(t) by the gain factor.
The processed forms of the input x(t) and y(t) are next processed by applying the well-known Hanning window. They are then transformed into their respective frequency domain using the well-known fast Fourier transform (FFT) and then to Bark Scales, P
x
(b) and P
y
(b), using the Bark Frequency Warping technique. The transfer function H(b) is then estimated using the Bark Scales. The transfer function is used to normalize P
y
(b), which, in turn together with P
x
(b), is used to estimate the gain G(b) which will be used to suppress the echo. Subsequently, the Bark Scales are unwarped and the gain function is then used to suppress the echo from the input x(t). The well-known inverse FFT (IFFT) and overlap add are performed to yield an echo-free signal.
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patent: 5588089 (1996-12-01), Beerends et al.
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Thomas W. Parsons, “Voice and Speech Processing,” McGraw-Hill, Inc., New York, 1987, pp. 29-39.*
M. R. Asharif, F. Amano, S. Unagami, and K. Murano, “Acoustic Echo Canceler based on Frequency Bin Adaptive Filtering (FBAF),” Proc. Globecom '87, 1987, pp. 1940-1944.*
Kazuo Murano, Shigeyuki Unagami, and Fumio Amano, “Echo Cancellation and Applications,” IEEE Communications Magazine, Jan. 1990, pp. 49-55.
Bitwave Pte. Ltd.
Ho Lawrence Y. D.
Korzuch William R.
Storm Donald L.
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