>>* Hi, *>>* *>>* I have a scenario in a mobile VoIP app that
requires echo cancellation but *>>* is somewhat different from what's
described in the docs. *>>* *>>* Audio is received from and sent to
the network at 8000Hz. Each packet *>>* contains 160 samples worth a
playback of 20ms. *>>* *>>* But the hardware requires aggregation
for both playback and capture. So for *>>* playback, I coalesce 4 packets
in a buffer and queue them as a larger buffer *>>* for playback.
*>>* On the send side, I read a large buffer (worth 4 packets) and send
them out *>>* over time 20ms apart. *>>* *>>* I tried using
speex_echo_playback just when a 160-sample packet arrives from *>>* the
network, before coalescing and speex_echo_capture just before a packet
*>>* is sent out to the network but that doesn't seem to work properly
(doesn't *>>* cancel any echo). *> The most likely reason is that
you didn't align the far-end and near-end samples.> So the filter can not
converge.
Thanks for your response. Can you please explain what you mean by
align samples from near-end and far-end? And how is that usually
accomplished?
>>* So, in this scenario above, please recommend a good place to insert
*>>* speex_echo_playback and speex_echo_capture. Should I be just before
the read *>>* and write to hardware? In that case, should I use a larger
"frame size" of *>>* 160 samples x 4? *> Of course you can
set frame size to 160*4. Otherwise you can feed samples 4 times> to the AEC
if you don't want to modify the frame size.
>* *>* Thanks in advance, *>* Daniel.** *
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