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...AEC works :) And I was referring speaker input as having direct access into what has been played thru speaker. (ASIO Soundcard provides interface for that) So if I am remebering correctly, with data from Mic and Speaker in sync, speex AEC should do the job. Tim _____ From: Guilherme Balena Versiani [mailto:guibv at comunip.com.br] Sent: Tuesday, August 11, 2009 1:33 PM To: Tim Na Cc: 'Greger Burman'; speex-dev at xiph.org Subject: Re: [Speex-dev] AEC troubleshooting Hello Tim, Please explain exactly what you are meaning by "expose speaker input as other OS". Did...

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...all back to a crude echo > suppression scheme. Skype seems to do a reasonable job, though. It is difficulty to measure the accurate difference between two sampling rate by a software in a short time. So I don't believe Skype do some kind of resynchronise or re-sampling work. From: Guilherme Balena Versiani <guibv at comunip.com.br> > 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 r...

...work. >> >They have plenty of time. It is possible to do the necessary resync >*provided* the PLL doesn't drift around too much when it is not being >driven by a proper source. Concern about that point might make them >avoid it completely. >> >> From: Guilherme Balena Versiani <guibv at comunip.com.br >> <mailto:guibv at comunip.com.br>> >> >> > 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 >> &...

Hello all, I'm just new to this list. I'm user of a voIP application which uses Speex Acoustic Echo Canceller (so only user of speex). The only parameter it has to tune is the length of the ec tail. I'm testing the echo canceller on an arm9 platform. Tests made only allow me to raise the length of the tail until 250 msec (upper the cpu power consumption is 100% and there are audio

Hi Andrey, May be that we don´t understand. I need to transform speex to pocket pc platform. The Windows API is different to Pocket Pc API. Some IDEs that help us to create code for pocket pc, for example, are Visual C++ 3.0 embedded or Visual Studio.Net. Christian will develop this work with Visual C embedded. If you can develop this work quickly or you can guide me step by step, i would be very

I'm being PHBed into a VOIP project, and Speex sprang to mind. Bandwidth is going to be a fairly serious issue for us. With regards to a Speex enc/decoder, I was wondering: Rick Kane and David Siebert have already asked about this, but seem to have gotten very different responses - the former a call to arms, and the latter a "well, if you do it, it'll get done." What's the

Then why ONE sound card have different capture and playback rate? It must be ONE single physical clock generator which is used by both ADC and DAC in the sound card, isn't it? If you are a hardware engineer. Will you design two different physical clock for ADC and DAC seperately? What on earth causes this problem? Who knows its intrinsic real reason? Isn't there any other solutions? For

Hello, Currently OpenSSH has a fixed order on how the key authenticates the user: at first it tries to authenticate against TrustedUserCAKeys, afterwards it does it against the output keys from the AuthorizedKeysCommand and finally against the files as set in AuthorizedKeysFile. I have an use-case where this order is not ideal. This is because in my case the command fetches keys from the cloud