2011/4/18 Li Maoquan <span dir="ltr"><<a href="mailto:limaoquan2000@126.com">limaoquan2000@126.com</a>></span><br><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div class="im">>>>> Hi, <br>>>> <br>>>> I have a scenario in a mobile VoIP app that requires echo cancellation but <br>>>> is somewhat different from what's described in the docs. <br>
>>> <br>>>> Audio is received from and sent to the network at 8000Hz. Each packet <br>>>> contains 160 samples worth a playback of 20ms. <br>>>> <br>>>> But the hardware requires aggregation for both playback and capture. So for <br>
>>> playback, I coalesce 4 packets in a buffer and queue them as a larger buffer <br>>>> for playback. <br>>>> On the send side, I read a large buffer (worth 4 packets) and send them out <br>>>> over time 20ms apart. <br>
>>> <br>>>> I tried using speex_echo_playback just when a 160-sample packet arrives from <br>>>> the network, before coalescing and speex_echo_capture just before a packet <br>>>> is sent out to the network but that doesn't seem to work properly (doesn't <br>
>>> cancel any echo). <br>>><br>>> The most likely reason is that you didn't align the far-end and near-end samples.<br>>> So the filter can not converge.<br>><br>>Thanks for your response. Can you please explain what you mean by <br>
>align samples from near-end and far-end? And how is that usually <br>>accomplished? <br><br></div>You need to know the total delay caused by DAC buffer before speaker, ADC buffer<br>after microphone and acoustic path between speaker and microphone. Simply to say,<br>
if you play an impluse signal and its first echo appears after N sample cycles,<br>you can call N as the delay between y (echo in near-end signal) and x (far-end<br>signal). Then you can buffer far-end signal for N-M cycles before sending to AEC.<br>
M is a little number (such as 100) in order to avoid filter failure when echo<br>path drifts.<div class="im"><br></div></blockquote><div> </div><div>Thanks again. I am trying to model the delay between the near and far end signals using a circular queue of length n. Every time a frame is received and queued for playback, it is also entered into the queue. Each frame being read from the mic is echo-cancelled ( speex_echo_cancellation ) using the oldest frame in the queue if the queue is filled up, thus I am cancelling the recorded frame using a playback frame that is N-frames old.</div>
<div><br></div><div>I have played with different values of N from 2 to 50 (320 samples to 8000 samples), attempting to align the input and output but the cancellation doesn't seem to work. The echo is steady as ever.</div>
<div><br></div><div>Is this model correct and expected to converge with a right value of "N"? Or do I need some other adaptation to account for drifts here. Right now, it's a black box for me. I am not sure how to get some feedback from this system to tune the AEC (and the delay parameters) correctly.</div>
<div><br></div><div>Also, I did not follow the use of "M" in your description above and how it helps with drifts. My queue stores frames (160 samples each). So a number of 100 samples seems too small.</div><div>
<br></div><div>Btw, I am assuming that speex AEC API can be used even though I am not using the speex encoder/decoder.</div><div><br></div><div><br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div class="im"><br>>><br>>>>> So, in this scenario above, please recommend a good place to insert <br>>>> speex_echo_playback and speex_echo_capture. Should I be just before the read <br>>>> and write to hardware? In that case, should I use a larger "frame size" of <br>
>>> 160 samples x 4? <br>>><br>>> Of course you can set frame size to 160*4. Otherwise you can feed samples 4 times<br>>> to the AEC if you don't want to modify the frame size. <br>>><br>
>>> <br>>> Thanks in advance, <br>>> Daniel.<br><br><br><br></div><span title="neteasefooter"><span><hr>
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