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Figure 10.20 Illustration of algorithm delay due to lterbank
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d(n) + Analysis filterbank P(z) Full-band cancelation Echo cancelation in subbands Analysis filterbank Synthesis filterbank
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Figure 10.21 Structure of delay-free subband acoustic echo canceler
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if the distance between the loudspeaker and the microphone is 1.5 m, the measured time delay in the system is about 4.526 ms based on the sound speed traveling at 331.4 m/s, which corresponds to = 36 at 8 kHz sampling rate. As discussed in 7, if a xed-point DSP processor is used for implementation and is suf ciently small, the excess MSE increases with a larger L, and the numerical errors (due to coef cient quantization and roundoff) increase with a larger L, too. Furthermore, roundoff error causes early termination of the adaptation if a small is used. In order to alleviate these problems, a larger dynamic range is required which can be achieved by using oating-point arithmetic. However, oating-point solution requires a more expensive hardware for implementation. As mentioned earlier, the adaptation of coef cients must be temporarily stopped when the near-end talker is speaking. Most double-talk detectors for adaptive line echo cancelers are based on ERL. For acoustic echo cancelers, the echo return (or acoustic) loss is very small or may be even a gain because of the use of ampli ers in the system. Therefore, the higher level of acoustic echo makes detection of weak near-end speech very dif cult.
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10.6.6 Testing Standards
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ITU G.167 speci es the procedure for evaluating the performance of an acoustic echo canceler. As shown in Figure 10.22, the echo canceler is tested with the input far-end signal Rin and near-end signal Sin , and the output near-end signal Rout and far-end signal Sout . The performance of echo cancelation is evaluated based on these signals. Some G.167 requirements are listed as follows:
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Initial convergence time: For all the applications, the attenuation of the echo shall be at least 20 dB after 1 s. This test evaluates the convergence time of adaptive lter. The lter structure, adaptation algorithm, step size , type of input signal, and prewhitening technique may affect this test. Weighted terminal coupling loss during single talk: For teleconferencing systems and hands-free communication, its value shall be at least 40 dB on both sides. The value is the difference between
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Simpli ed diagram for G.167 testing
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the signal level (in Sout ) without echo cancelation and the signal level with echo canceler in steady state. Test signal is applied to Rin and no other speech signal other than the acoustic return from the loudspeaker(s) is applied to the microphone.
Weighted terminal coupling loss during double talk: For teleconferencing systems and hands-free communication, its value shall be at least 25 dB on both sides. After the echo canceler reaches the steady state, a near-end speech is applied at the Sin for 2 s. The adaptive lter coef cients are frozen and then the near-end speech is removed. This test evaluates how fast and accurate is the DTD to stop the coef cient update during the double talk. Recovery time after echo path variation: For all the applications, the attenuation of the echo should be at least 20 dB after 1 s. This test evaluates the echo canceler after the double talk; the coef cients may be affected but the system should not take more than 1 s to update to the optimum level.
One of the interesting observations for the tests speci ed by ITU-T G. 167 is that the test vectors are arti cial voices according to ITU P.50 standard. These arti cial voices are composed using the speech synthesis model. This makes the test easier with reduced limitation of human resources.
10.7 Experiments and Program Examples
This section presents some echo cancelation modules using MATLAB, C, or C55x programs to further illustrate the algorithms and examine the performance.