similar to: Playing captured speex frames

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

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:

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

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

Hi, Searched far and wide for a solution to this problem but couldn't find anything. I'm debugging a VoIP application that uses Speex and need to playback the raw speex frames to verify the quality. Wireshark helped me save the RTP payloads into a file but unfortunately, the speexdec utility seems to be wanting a valid header at the start of the file without which it reports an error.

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. -

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

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 36 and 32 bytes. A five line perl script will do it. - farhan On 7/7/11, Vinod Panicker <vinod.p at

I am doing this right now with no problem. I am not using speex for this at the moment though. Group delay is the biggest problem. I implemented a version where the input and output sample rates are known up front. The routine than interpolates between the jitter. This should solve the problem. The crystals used to clock the input and output have very fine tolerances on most standard audio

Seems to me that they are using speex already. Ethereal shows that the voice RTP stream's payload type is 103. According to this page [ http://www.openh323.org/pipermail/openh323/2004-June/068705.html], 103 is SpeexNarrow-8k, although this email from Craig might not be correct. - Cheng > Message: 2 > Date: Thu, 25 Aug 2005 04:15:44 -0500 (CDT) > From: Ashhar Farhan

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

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

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

Hi Shridhar, Sample rate conversion is not enough to solve this problem. I have tried this method several months ago. The first step is to measure the difference between sample rate of capturing and rendering. Then resampling (by what you said "sinc interpolation") one signal to eliminate the difference. The frequency step in my experiment is less than 0.1Hz. I have tried speex AEC

On 04/13/2011 02:58 AM, Shridhar, Vasant wrote: > I am doing this right now with no problem. I am not using speex for this at the moment though. Group delay is the biggest problem. I implemented a version where the input and output sample rates are known up front. The routine than interpolates between the jitter. This should solve the problem. The crystals used to clock the input and

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, i am trying to compile a fixed point version of lipspeex on desktop windows environment. a short reason for the complicated story is that we need to keep the 'engine' part of our project same across the various ports of our voip project (blame the pointy haired boss). now, when i compile with FIXED_POINT defined in the Visual C++ 6.0, i get these errors:

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

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