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...back Actually, I just tried again with the wav file directly and it just seems to be a bug with the way the speexenc utility handles wav files. The problem doesn't occur with the equivalent raw file (which I used for testing). It's probably really stupid anyway. Jean-Marc Le mercredi 06 juillet 2005 ? 14:00 +0300, David Resnick a ?crit : > I am using speexenc/speexdec to convert WAV files to Speex and then > play them back. At the end of playback for some files there is a tick > sound heard that isn?t in the original WAV file. > > > > I have no prior experience wi...

hi, guys. I come from china, a student of BUPT. I have developed a softphone using osip&ilbc. Now, i want to add speex to my softphone. I heard speex have been a daft of rfc, but I just found draft of the format of rtp in ietf's website. Is speex dead? Who can tell me that? Thanks in advanced. yunyong

❦ 25 juillet 2016 06:06 CEST, Martin Kletzander <mkletzan@redhat.com> : >>╭─────┤ http://libvirt.org/formatnetwork.html#elementsConnect ├───── >>│Inclusion of the forward element indicates that the virtual network is >>│to be connected to the physical LAN.Since 0.3.0. The mode attribute...

❦ 25 juillet 2016 08:44 CEST, Martin Kletzander <mkletzan@redhat.com> : >>> "Just a vnet interface, no bridge" means you want no network. Where >>> should the vnet be connected? >> >>The host is able to handle the routing. It seems I could just declare an >>...

Hi all, The problem is on the volume of the voice sent by the SPA-841. I think the echo cancel algorithm sets a limit to the microphone when detects sounds or noise from the earphone. This problem generates an oscillation on the voice volume sent by the phone and even turns it off completely for very little lapses of time making the communication very uncomfortable. I manage three different

Hi List; How can one Asterisk Box A to send a SIP call for another Asterisk Box B, and that call to be authorized based on the username and password, and not on the IP (as the IP address of the source is not known because it keep changing)? I think the trick in the Dial command, how to write it properly in a way that other Asterisk Box can recognize the sip username and password which are existed

Hello All I installed shorewall 3.0.8 on Centos 4.3 with openvz.org kernel it work well i have in this Host 3 virtual servers (VPS) i can access from a VPS to the internet , and with NAt rule (Via Shorewall) i can access from Internet to the 3 VPS. i want that all the 3 VPS can communicate between them. i can''t do a tcp connection from a VPS to an other , in my shorewall log in the

Hi people, As I said before: I did the IDCT to run on the FPGA. My friends from university did the Reconstruction routines running on the FPGA. I'm helping with the LoopFilter, and it is almost there. (all VHDL) I did a small profiling of the libTheora running on a Altera Stratix II device: The processor used was the NIOS II with 8Kb of data and instruction cache, branch prediction and

.... The packet loss concealment is also done, but not the VAD/VBR. The jitter buffer isn't converted, but there's not many float ops there in the first place. The preprocessor and echo cancellor aren't converted and there's lots of float stuff there. Jean-Marc Le mercredi 20 juillet 2005 ? 09:18 -0700, gue baja a ?crit : > Is there any place where I can see a summary of what > is being done and what is still pending with the fixed > point version of the libraries? > > I have some experience with vocoders and fairly good > experience with de-jitters and suchs....

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: good.wav Type: audio/wav Size: 7430 bytes Desc: not available Url : http://lists.xiph.org/pipermail/speex-dev/attachments/20050706/a77cade6/good-0001.bin -------------- next part -------------- A non-text attachment was scrubbed... Name: bad.wav Type: audio/wav Size:

Le vendredi 30 juillet 2010 12:25:48, Michael Smith a ?crit : > All that said: Icecast2 is largely unmaintained these days - I don't > know if anyone is interested in going through these and figuring out > which ones are mergeable, which need fixing, and which shouldn't be > used at all. Maybe its ti...

Hi there, here is a little bug report: # ./configure --with-gnu-ld --enable-sse # make [...] vq.c:99: error: conflicting types for `vq_nbest' vq.h:44: error: previous declaration of `vq_nbest' vq.c:133: error: conflicting types for `vq_nbest_sign' vq.h:46: error: previous declaration of `vq_nbest_sign' The --enable-sse option took this bug to the surface. The header file is

If you have a couple minutes, it probably wouldn't be hard to see where the bug is (I just don't have time this week). It would have to do with the granulepos computation in speexenc. Jean-Marc Le jeudi 07 juillet 2005 ? 08:41 +0300, David Resnick a ?crit : > Thank you very much for your immediate attention! > > I've exported my wav files to raw files before encoding and the click is now > gone. That sure beats discarding the last packet, which was the workaround > "solution" I...

_____ De : Pousset [mailto:maud.pousset@noos.fr] Envoyé : mardi 4 juillet 2006 18:38 À : 'r-help@stat.math.ethz.ch' Objet : Latent Class Analysis Hello everybody, I am working on latent class analysis and have already used the ‘R’ function « lca » (in the e1071 package). I ‘ve got interesting results but I can’t simply find out the methodology used by thi...

Le mercredi 19 juillet 2006 ? 20:05 +0800, Janus.Wang@QMITW.COM a ?crit : > Hi Jean-Marc, > Please read the attachment file for Marvell 8618 datasheet for your reference, Too much is like not enough. Care to summarize? (speed, arch level, ...) > BTW, Could you mind to specify the voice computing power for Spee...

ok, sorry, my mistake was the C printf. Thank you for your good answer Regards ----- Message d'origine ---- De : JS Ubei <jsubei at yahoo.fr> ? : jim holtman <jholtman at gmail.com> Cc : r-help at r-project.org Envoy? le : Jeudi, 17 Juillet 2008, 15h25mn 07s Objet : [R] Re : float and double precision with C code thank you for your quick answer, I'm far of the digits capacity and my values are not the result of a computation. I'm developping a R package to acces a specific data source. And I need precision a few better. How...

Hi, When a want to use scheduled_task for create a windows task and when i give a AD user, i have this message : err: /Stage[main]/Mirnas/Scheduled_task[test]/ensure: change from absent to present failed: Invalid user: DOMAIN\oper My user domain and the password are correct. I have the same error if i change DOMAIN\oper by oper@domain.com or DOMAIN/oper With a local user, is working, but i

Probably the level of your signal is too low and/or you're just not letting it time to adapt. Jean-Marc Le mercredi 19 juillet 2006 ? 19:00 -0400, ac2491@columbia.edu a ?crit : > On closer looks and debugging I always end up in > > speex_echo_cancel function with comment > /* Temporary adaption rate if filter is not adapted correctly */ > > > Does this give any clue to the problem? I wonder why it w...

Hello ! I am developing a kind of phone using TI's OMAP 5912 and speex. The soft works great on ARM, and it is time to move speex and voice processing stuff to DSP. Could some one point me how to do it ? Should I install CCS ? Or it possible to compile speex using just ti_dsptools ? Does anybody ran speex and dsp gateway ? It would be great, if some one share makefile, cmd etc. for