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...--enable-shared --enable-threads=posix --disable-checking --with-system-zlib --enable-__cxa_atexit --host=i386-redhat-linux Thread model: posix gcc version 3.2.3 20030502 (Red Hat Linux 3.2.3-52) [guor@ppe speex-1.2beta1-install]$ make make all-recursive make[1]: Entering directory `/home/guor/osip/dev/speex-1.2beta1-install' Making all in libspeex make[2]: Entering directory `/home/guor/osip/dev/speex-1.2beta1-install/libspeex' make[2]: Nothing to be done for `all'. make[2]: Leaving directory `/home/guor/osip/dev/speex-1.2beta1-install/libspeex' Making all in include make[2]:...

...re all 3 part of a minimal list (draft-ietf-avt-rtp-speex at tools.ietf.org) dedicated to latest discussion about the draft. I have answered some questions, but there are small changes and adaptation still required to the ietf draft... tks, Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 - http://savannah.nongnu.org/projects/exosip/

...ile it with: > -D_USRDLL > > I'm not able to understand completely why: anyway, it > now can be compiled without breaking anything. > > I'll now test wether the aec have an impact or not :) > tks, > Aymeric MOIZARD / ANTISIP > amsip - http://www.antisip.com > osip2 - http://www.osip.org > eXosip2 - http://savannah.nongnu.org/projects/exosip/ > > > On Mon, 6 Nov 2006, Aymeric Moizard wrote: > >> >> >> On Mon, 6 Nov 2006, Jean-Marc Valin wrote: >> >>> That's quite a strange problem. Considering that Speex...

...ean-Marc, > > I remember you said that the SDP tells the mode information. Right. But a decoder like speex can always be configured from information contained in the incoming stream. That makes your application more interroperable. Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 - http://savannah.nongnu.org/projects/exosip/ > Is SDP what you called in-band signalling? > Lianghu > > On 12/6/06, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: >> >> In a "normal" frame that doesn't have in-b...

...> I want to be able to automatically detect the mode (narrowband/wb..) > of the RTP stream I'm receiving. > > I guess this information is included in frames? Isn'it? > How can I read it? > > tks, > Aymeric MOIZARD / ANTISIP > amsip - http://www.antisip.com > osip2 - http://www.osip.org > eXosip2 - http://savannah.nongnu.org/projects/exosip/ > > _______________________________________________ > Speex-dev mailing list > Speex-dev@xiph.org > http://lists.xiph.org/mailman/listinfo/speex-dev > >

...sh to test another application using speex to prove myself my >> application has a bug: if somebody has one, please send it. >> >> Sorry for bothering you with probably non-speex problems, >> >> Aymeric MOIZARD / ANTISIP >> amsip - http://www.antisip.com >> osip2 - http://www.osip.org >> eXosip2 - http://savannah.nongnu.org/projects/exosip/ >> >> _______________________________________________ >> Speex-dev mailing list >> Speex-dev@xiph.org >> http://lists.xiph.org/mailman/listinfo/speex-dev >> >> > >

...omponent. > > Note that the input has really low volume which is most > probably the main issue here. Still this is showing that > using the AGC potentially introduce DC component? Any > comment? > > Tks, > Aymeric MOIZARD / ANTISIP > amsip - http://www.antisip.com > osip2 - http://www.osip.org > eXosip2 - http://savannah.nongnu.org/projects/exosip/ > > _______________________________________________ > Speex-dev mailing list > Speex-dev at xiph.org > http://lists.xiph.org/mailman/listinfo/speex-dev > >

...hey met such issue and solved it? Just in case... I also wish to test another application using speex to prove myself my application has a bug: if somebody has one, please send it. Sorry for bothering you with probably non-speex problems, Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 - http://savannah.nongnu.org/projects/exosip/

Hi, Speex-dev: ? I have some problems about complie speex on pxa-270. I use libspeex.so.1.4.0 at my platform after complie speex and all function is ok, but the cpu loading is 99%. How can i do to reduce the cpu loading? ? note: 1. My Platform are os: linux 2.6.19 , cc: gcc-2.95.3 , cpu: pxa-270, ogg: libogg-1.1.3, speex: speex-1.2beta3 2. I configure speex by the follow command. ./configure

Sorry, got the subject wrong in the last mail. My e-mail client is playing games with me...

...rary is compiled with: -D_LIB While I should compile it with: -D_USRDLL I'm not able to understand completely why: anyway, it now can be compiled without breaking anything. I'll now test wether the aec have an impact or not :) tks, Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 - http://savannah.nongnu.org/projects/exosip/ On Mon, 6 Nov 2006, Aymeric Moizard wrote: > > > On Mon, 6 Nov 2006, Jean-Marc Valin wrote: > >> That's quite a strange problem. Considering that Speex doesn't have any >> "init code...

Dear Sir/Miss: I'm very sorry to bother you,but I've found a bug in the Speex,Version 1.2beta1,and also I've have some questions. I'm deeply impressed by the work you have done in the coding and decoding of speech signal.But when using speex,I encoutered a bug,that is when in 32K mode,the window size is still 200 samples,but not 400 samples,the bug will bring the problem of

...IP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1571536832 From: <sip:203@192.168.1.100>;tag=1543062237 To: <sip:203@192.168.1.100>;tag=1543062237 Call-ID: 2259600998@192.168.1.100 CSeq: 0 REGISTER Contact: <sip:203@192.168.1.100> max-forwards: 10 expires: 900 user-agent: oSIP/Linphone-0.12.1 Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.1.100 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1571536832 From: <sip:203@192.168.1.100>;tag=1543062237 To: <si...

Aymeric Moizard a ?crit : > > I'm working on the echo canceller with the SVN, with Windows Mobile 5 > PocketPC. > > Obviously, I can't make it run correctly. What happens/doesn't happen? > My question: If I use the SPEEX_PREPROCESS_SET_ECHO_STATE, I have > changes in the sound: noise seems to be removed. Noise should be removed even without

Hi We are working on a speaker phone system using PJSIP and Speex Speech processing API on an ARM platform. Currently we have spent about a month on getting the AEC to work properly and we have worked through the most common causes of problems (such as clock drift, synchronization problems and non-linearity's in echo path). Now we achieve ERLE of about 17 dB which tells me that the AEC is

...ans that the echo canceller needs to re-adapt. So as soon as the (local) clocks aren't *perfectly* synchronised, the echo cancellation performance goes down to a point where it's mainly unusable. Jean-Marc > Tks, > Aymeric MOIZARD / ANTISIP > amsip - http://www.antisip.com > osip2 - http://www.osip.org > eXosip2 - http://savannah.nongnu.org/projects/exosip/ > > > On Wed, 1 Nov 2006, Tom Grandgent wrote: > >> Isn't this the same problem described starting at the bottom of >> this page? >> http://www.embeddedstar.com/articles/2003/7/art...

Isn't this the same problem described starting at the bottom of this page? http://www.embeddedstar.com/articles/2003/7/article20030720-11.html Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > > > As it says in 5.4.1 of the good book "Using a different soundcard to do > > the capture and playback will *not* work, regardless of what you may > > think.

What does this mean? Asterisk 1.2.13 talking to 1.4.0. (response from 1.4.0.) Yuan Liu

...arly to cope with packet loss or delayed initiation (initial packets are often lost at the beginning on one side of the conversation). I think this approach will fit your need for keeping CELT as low as possible but mandatory for VoIP. tks, Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 - http://savannah.nongnu.org/projects/exosip/

Does anybody know of a Speex / RTP implementation in C or Java ? Thanks,