similar to: speex 1.1.8

Hi, Ok, what you suggest sound logical to me. Currently, I have done a small trick to prevent this problem. What I did is that whenever windows request a voice packet from me and if I do not have the voice packet, I repeat the previous packet. Hence, all the breaking portion is filled with previous packet. This trick seems to work so far. I am not sure what is the side effect. I think jitter

Hi everyone, As many noticed, I just released 1.0.5 and 1.1.8 over the weekend. There isn't much new in 1.0.5, except that I added all the API changes from 1.1.x so that it's possible to write code that will work with both 1.0.5 and 1.1.8. As for 1.1.8, here are the changes: - Initial TI C5x port: I integrated a patch from Jamey Hicks - Some fixed-point improvements and fixes - Better

> Ok, what you suggest sound logical to me. Currently, I > have done a small trick to prevent this problem. What > I did is that whenever windows request a voice packet > from me and if I do not have the voice packet, I > repeat the previous packet. Hence, all the breaking > portion is filled with previous packet. This trick > seems to work so far. I am not sure what is the

On Tue, 11 Oct 2005 10:36:51 +0200, Jean-Marc Valin <Jean-Marc.Valin@USherbrooke.ca> wrote: > Source code at: > http://people.xiph.org/~jm/speexclient/ I rewritten my program so it is more similar to yours, I grabbed your code for storing and retrieving echo, and I have better results, I think the echo is cancelled in about 50% but it still can be heared. I think there is a

Hi, When an audiopacket is received too late I could interpolate this frame. The problem is that I don't know if it is a true bufferloss or just the last audiopacket of a talkspurt. Now my question is if it's harmfull for the audioquality that at the end of the talkspurt one frame is interpolated? Or would this be almost inpossible to hear since the last audiopacket in the talkspurt

Hi, I implemented the echo cancellation from speex in my Windows VOIP-client VoIPerized --> www.voiperized.com Greetings Jeroen de Kleijn -----Original Message----- From: speex-dev-bounces@xiph.org on behalf of hs Sent: Tue 10/11/2005 4:02 PM To: speex-dev@xiph.org Cc: Subject: Re: R: [Speex-dev] echo cancellation On Tue, 11 Oct 2005 10:36:51 ?, Jean-Marc Valin

Hi, I not really keeping track of the revision numbers for releases. As for the tags, 1.1.8 is at http://svn.xiph.org/tags/speex-1-1-8/ and 1.0.5 is at http://svn.xiph.org/tags/speex-1-0-5 For older releases, the tags are sometimes just rel-x-y-y because that's what I was doing before the switch to svn (the tree was imported from cvs). Oh, and the 1.0 branch is in

Hi, I've been getting reports of stuttering audio in my VoIP application that uses Speex 1.1.11. Since I haven't had this report with previous versions of Speex I was wondering if one of the bug fixes in 1.1.12 solves a bug which causes the stuttering. To me the stuttering report sounds like the decoder state that speex uses gets messed up. Some info, I use the floating point version of

Hi Jean-Marc, For the residue in the echo_cancel function I pass in a pointer to an array of floats. First I used an array the size of the framesize, but I discovered this causes crashes because the echo_cancel funtion writes framesize+1 floats to the pointer. I find this plus one strange, since almost all things are exactly the framesize. Is this an error in the echo_cancel function or simply

Hi, In the past somebody posted a patch on the list to choose the SSE at runtime using a function called speex_set_asm. I have been searching the archives, but I could not found that specific post. I noticed the new 1.2beta1 does not include this patch, so I was curious if the person who originally made this patch has made the same modifications to the new 1.2beta1. Greetings Jeroen de

Le jeudi 03 mars 2005 ? 20:29 +0100, Kleijn, J.S. de a ?crit : > I use the Intel compiler 7.0 , and it doesn't like it Does it have an option to enable C99? Otherwise, if you're aware of another mechanism that allows it to allocate temporary stuff on the stack, I might be able to support it. Jean-Marc > -----Oorspronkelijk bericht----- > Van: Jean-Marc Valin

> - set sampling rate to 8 kHz (at least for now) > - make sure the far end signal in the playback signal is always a bit in > advance (never late) compared to the mic signal. > - Set the tail length to something around 100 ms. > > Also, if you're using two different soundcards (as I understand) for the > playback and the capture, you're *never* going to get echo

Concerning the VAD in SPEEX, I've been using "VoIPerized" a phone using SPEEX sampling at 16K, I use a Plantronics USB headset. I find that I have to set the VAD quite sensitive in order not to lose "S's" from the start of a word/sentence. the result is that I'm often sending packets unintentionally resulting in packet loss from the incoming stream(usually caused

Anyone know if it is possible to control how aggressively the "Aggressive" mode behaves. Meaning, is it possible to dial back the aggressive mode to have a happy medium between Regular and the Aggressive defaults. I have a situation where Normal echo cancellation is not quite enough, however when I turn on aggressive mode We are attacking it to hard and I am unhappy with the walkie

Hello everybody, I'm glad to announce the major release of xocc C compiler, 0.9.2, based on BSD license. xocc is a C compiler that based on XOC infrastructure, XGEN code generator and XOCFE C frontend. Both xocc, XOC, XGEN, and XOCFE are based on BSD license. XOC provides multi-level operations, flexibility, and the capability of representing almost all popular languages. There

Hi all, I was excited to see the announcement on the list regarding the fix for the echo problems on Digium FXO cards! I have 2 X101P's, TDM400P with 4 FXS modules and couple of XLite softphones. A few months back,I had gone thru the recommendation on the list to remove echo from the SIP phones(I never did have any echo on the TDM400P FXS phones), and had removed about 90% of the echo.

I'm trying to compile 1.1.8 with dovecot-1-1-989789b2afd8 patches applied on Solaris 10 sparc using gcc 2.95.3 and also tried version 3.4.3. This is error I'm getting: gcc -DHAVE_CONFIG_H -I. -I../.. -I../../src/lib -I../../src/lib-auth -DPKG_RUNDIR=\""/userM/mail-services/dovecot/var/run/dovecot"\"

Hello! I think,what nobody discuss theme with this programme. I have decided to update the program, because this program is not starting. The version wine 1.1.8, Voicecheap was installed and fast started without problems.But then I have seen that it does not accept my login and does not accept connection to Voip server.At attempt to enter to the settings - program does not answer. If someone could

Hello I noticed a problem after upgrading to 1.1.8 Users that access to their mailboxes using webmail squirrelmail and HTTPS using HTTP seems to work correctly but it not safe enough ... There is an error message saying the imap server send a BYE command ... I downgraded our Dovecot server to 1.1.7 then it works again ... Does somebody has the same problem ? -- Cordialement Frank Bonnet

Hi, We're seeing a new error under Dovecot 1.1.8 that we haven't seen before. Sep 14 09:00:10 hostname dovecot: [ID 684838 local6.error] child 638 (imap) killed with signal 10 This is being seen under dovecot 1.1.8, on Solaris 10u6. This is the only server we've got running this software. We're not seeing this on any of our other 5 servers running the 1.1.x series. Namely