Another important thing is the external echo path (e.g. outside the phone's case). The external echo path is actually a number of different echo paths due to the room objects reflecting the speaker's sound back to the microphone. It has also been noted that these echo paths vary with time as the objects or people move in the room. The changes in the echo path impulse response cause an increase in the residual echo error signal. This forces the AEC to start adapting to the new impulse response and it can even diverge, if the changes are fast or abrupt. In the installed phone, the speaker and microphone should not be directed to the path that is subject to fast changes. It is usually better to direct the speaker and microphone towards the ceiling since this echo path changes rarely.
Besides the main hardware questions like the nonlinear distortions, there are also certain requirements on the signals, which are fed as samples to the echo cancellers.
The first requirement is that signal delays in software be as short as possible. In general, there should be no signal processing done between the codec and the echo canceller. And there must be no sample accumulation without any good reason for it. Excessive buffering will increase the effective signal delay in the echo path and therefore the utilization of the filter coefficients will be ineffective (some of the coefficients will have to cover the additional delays yet they will be zeroes). Obviously, the reference signal, y(i), delay in software must be smaller than or equal to the software delay for the signal with echo, x(i)+r(i). If this is not true, the echo canceller will not be able to converge and cancel the echo since it doesn't have the reference signal, which is to be subtracted.
Attention must be paid to the signal delays in the software in another respect. All of the delays must be constant throughout the entire session, in which echo cancellation is desirable. Changing the delays during a phone call will cause the echo canceller to diverge and stop canceling the echo until it converges again.
It is also possible to have other problems with signals in software. Echo cancellers usually process linear PCM samples, while the signals in memory or received from the codecs may be compressed to A-law or m-law samples. Make sure the echo canceller is receiving the samples in the format, which it was designed for. And don't artificially clip the samples on the way between the echo canceller and codec. This all will only contribute to the undesired nonlinear distortions.
The last but not least problem with echo canceller integration can be again due to the hardware or software design mistakes. What is the problem of incorrect codec synchronization? Well, the problem is easy to understand and relatively easy to solve, provided we know the right solution to it.
Suppose we have a device, which has several different signal sources, each clocked at a different rate, and the signals from one must go through the device to the other. Where is this possible? This is possible in hands-free phones, which have a pair codecs. One of the codecs is used to interface to the phone line and the other one is used to interface to the loudspeaker and microphone.
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