similar to: Re: Speex-dev Digest, Vol 15, Issue 15

Speexdec plays ogg files. You can simply modify it to play your raw file or even rtp capture. Jozsef -----Original Message----- From: speex-dev-bounces at xiph.org [mailto:speex-dev-bounces at xiph.org] On Behalf Of speex-dev-request at xiph.org Sent: Saturday, July 09, 2011 12:00 PM To: speex-dev at xiph.org Subject: Speex-dev Digest, Vol 86, Issue 3 Send Speex-dev mailing list submissions to

I'm trying to use a Howlink CL-100 ip phone with * It's h323 phone with very limited protocol support. But it's enough that I can use it to dial netmeeting client and artisoft pbx just fine. When I try to dial my * with it using either chan_h323 or oh323, it seems to fail on negotiating H245. Maybe this phone doesn't support it? I've used all different versions of

vincent, it should work out of the box. real time speex should work on the iphone as it has a really fast arm cpu (and speex works in realtime on even the 201mhz arm cpus). while compiling, remember to define the fixed-support support and you are all set. narrow band (8khz) and 8kbps bitrate will work very well. while you are at it, pse send the xcode project file to include in the tarball. -

hi .. i've just installed and licensed an instance of the G729 codec. I am trying to connect through asterisk to Go2Call server .. According to their info it involves dialling extension 729 on voip01.go2call.com, to get the IVR. my extensions.conf shows : exten => s,2,Dial(OH323/h323:729@216.52.153.206) which I think is correct, I have G729 enabled in the OH323.conf file and it seems to

Vincent Burel a ?crit : > ok, interesting ! > Do you know how much CPU it takes for a real time encoding/decoding ? it's > more around 20% or 80% instead ? > Do you know if it works for 44100 Hz / 32kbps / Quality =8 ? Did I ever mention that 44.1kHz is *not* recommended. Usually, 16 kHz is best, although 32 kHz can be good in *some* cases. In any case, you'll need to listen

ok, interesting ! Do you know how much CPU it takes for a real time encoding/decoding ? it's more around 20% or 80% instead ? Do you know if it works for 44100 Hz / 32kbps / Quality =8 ? thanks again. Vincent ----- Original Message ----- From: "Ashhar Farhan" <farhan at phonestack.com> To: "Vincent Burel" <vincent.burel at vb-audio.com> Cc: <speex-dev at

Hello Ashhar, thanks for your reply, but i'm a bit surprised. 1- Does it mean that it is possible to make a Real Time Encoding/Decoding with SPEEX on iPhone ? 2- What parameters have been used (QUALITY, SAMPLING_RATE, BITRATE) ? and what kind of CPU load do they measure for that ? Thanks by advance, Vincent ----- Original Message ----- From: "Ashhar Farhan" <farhan at

On Wed, 2006-06-28 at 11:34 +0530, Ashhar Farhan wrote: > I have sent the visual studio project files that compile the speex > client to jean-marc directly (as an attachment), sometime this week, I > will also upload the binary builds of static library files to my site > www.phonestack.com. Actually, Zen (illi on IRC) is the one handling the Windows stuff. I can't do anything with

Why don't you just try it? From what others have been reporting, it shouldn't take you long to get it running. You can use speexenc and speexdec for testing. On Nov 12, 2008, at 2:26, "Vincent Burel" <vincent.burel at vb-audio.com> wrote: > ok, thanks for these precision, and do you have some measure about > CPU load > ? > i really would like to get a

What frame size are you using? Jean-Marc Ouss a ?crit : > Hello, > > I have an (Access violation reading location) on line 115 in the fftwrap.c : > out[i] = scale*in[i]; > > with i = 1193 > > Only with preprocessor active > Systems : Visual studio 2005, XP, WB 16000 Hz, last svn. > > Hope it helps you. > > Ouss > > > -----Original

thanks for your reply, could you explain why 44.1kHz is *not* recommended with SPEEX ? Vincent ----- Original Message ----- From: "Jean-Marc Valin" <jean-marc.valin at usherbrooke.ca> To: "Vincent Burel" <vincent.burel at vb-audio.com> Cc: "Ashhar Farhan" <farhan at phonestack.com>; <speex-dev at xiph.org> Sent: Wednesday, November 12, 2008

Speexdec is expecting the frames to be wrapped in an Ogg frame I believe. Since you have the raw data can't you just add the Ogg header information to this before passing it to Speexdec? Vas ________________________________________ From: speex-dev-bounces at xiph.org [speex-dev-bounces at xiph.org] On Behalf Of Vinod Panicker [vinod.p at gmail.com] Sent: Friday, July 08, 2011 12:13 AM To:

On Fri, Jul 8, 2011 at 6:39 AM, Ashhar Farhan <farhan at phonestack.com> wrote: > what you need to do is this: take the wireshark raw dump, read each > udp packet and write it back to another file. While writing back to > the new file skip the sizeof udp header + rtp header. I can't recall > how many bytes you need to skip, however, I suppose it would be in the > range of

Can you provide me with a backtrace? It's quite hard to see what could happen. Also, can you reproduce the problem with testdenoise or only with your own code? Jean-Marc Ouss a ?crit : > The frame size is 320 > > Ouss > > > > -----Original Message----- > From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] > Sent: lundi 23 octobre 2006 18:26 > To:

Hi Farhan, My mistake, it is 9.6kps because is on gsm ! Regards, Danny -----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On Behalf Of Danny Chan Sent: Friday, November 19, 2004 2:08 AM To: Ashhar Farhan; speex Subject: RE: [Speex-dev] jitter buffer Hi Farhan, It is interesting that GPRS is 9.6 kps, as per GPRS standard it should be 115kps ? if

The frame size is 320 Ouss -----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: lundi 23 octobre 2006 18:26 To: Ouss Cc: speex-dev@xiph.org Subject: Re: [Speex-dev] Bug with the new preprocessor What frame size are you using? Jean-Marc Ouss a ?crit : > Hello, > > I have an (Access violation reading location) on line 115 in the fftwrap.c

Hello, I have an (Access violation reading location) on line 115 in the fftwrap.c : out[i] = scale*in[i]; with i = 1193 Only with preprocessor active Systems : Visual studio 2005, XP, WB 16000 Hz, last svn. Hope it helps you. Ouss -----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On Behalf Of speex-dev-request@xiph.org Sent: dimanche 22 octobre

Hi Farhan, It is interesting that GPRS is 9.6 kps, as per GPRS standard it should be 115kps ? if it is 9.6kps is true then I would use speex codec on GPRS for my project, since in some country they charge GPRS on flat rate rather than expensive airtime. Thanks for your work ! Regards Danny -----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On

Hi Farhan, What is the purpose of the protocol? Just to be a binary form a SIP? I mean conferencing, PTT, chat we have in h.323, sip, jingle, iax? What is the reason for a new one? The NAT traversal has been fixed for a while for all those protocols also. Diana farhan@phonestack.com wrote: > Dear All, > > I have finished documenting my Lightweight Telephony Protocol. It is > now

Hi Ashhar, could you please create a patch (in `diff -u` format) that fixes the build on symbian (preferably without breaking any other platforms ;-) and send it to the list? Even better, if you are actively maintaining code that uses speex on symbian, perhaps you could be given an svn account to check the changes in directly (if Jean-Marc is happy with that). Either way, it's best if you