Yeah, the ADC and DAC work at different rates. I really don't know why these soundcards are designed this way, but I can tell you that this is very common. In fact, I didn't ever find a soundcard with the same capture and render rates.<br>
<br>If you want to sinchronize capture and render parts, you need to implement a kind of buffer control, referenced as "skew control" (check it out). There is a very common control that removes or increases the delay when there is not enough energy on the output stream (i.e. the voice being acquired by the microphone).<br>
<br>But these kind of controls seem to put AEC of Speex crazy... When the skew control occurs, normally the AEC stops to work fine.<br><br>Regards,<br><br>--<br>Guilherme Balena Versiani.<br><br><br><div class="gmail_quote">
2010/6/9 limaoquan2000 <span dir="ltr"><<a href="mailto:limaoquan2000@126.com">limaoquan2000@126.com</a>></span><br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Then why ONE sound card have different capture and playback rate?<br>It must be ONE single physical clock generator which is used by both ADC and DAC<br>in the sound card, isn't it?<br>If you are a hardware engineer. Will you design two different physical clock for<br>
ADC and DAC seperately?<br>What on earth causes this problem? Who knows its intrinsic real reason?<br><br>Isn't there any other solutions?<br>For example, I have also tested Microsoft MSN and Skype. It seems that AEC<br>
function of those softwares are all no problem. Their AEC are designed by GIPS.<br>I guess that must be some solutions.<br>The solution of GIPS is only echo suppression?<br><br>Em 02-06-2010 04:41, <a href="mailto:ce09@polyu.edu.hk" target="_blank">ce09@polyu.edu.hk</a> escreveu:<br>
> Hi All,<br>><br>> I am a research associate in the Hong Kong Polytechnic University. One of<br>> my research interests is acoustic echo cancellation. Now I meet a big problem.<br>> When I was testing my own AEC module, I found that it was almost perfect in<br>
> few computers but much worse in other computers. Then I tried AEC module in<br>> speex, it was almost the same result.<br>><br>> Then I found the reason is that sound cards of most computers have different<br>
> sampling and playing rate, so it is almost impossible for the adaptive filter<br>> to converge and reach best result. For example, long time average sampling and<br>> playing rate of a sound card is 7999.864Hz and 7980.129Hz respectively. Sound<br>
> cards of the computers which shown perfect AEC result are the only sound cards<br>> which keep the same sampling and playing rate.<br>><br>> Then, what can I do to solve the problem? Could you please give me any advice?<br>
> BTW: I am working on Windows XP platform and I have tried DirectSound.<br>><br>> Your Sincerely<br>> Li Maoquan<br><br><br><br><span title="neteasefooter"></span><hr>
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