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But that's what a jitter buffer will do, eventually - drop a packet. And it's better to drop a silence packet. Cheers, .a -----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: Wednesday, December 21, 2005 3:39 PM To: Alex Bakaev Cc: Shantanu Gramsci; speex-dev@xiph.org Subject: RE: [Speex-dev] Bitrate problem On Wed, 2005-12-21 at 15:29 -0800,

[Just curious, and seizing the opportunity to communicate with other folks who are doing the same kind of thing I am...] How are you measuring the latency? I tried measuring it with my program (also Win32-based, also using DirectSound[Capture]) and came up with around 130ms. To measure it, I placed the mic near a speaker to get feedback going, had my program connect to itself (local

In my experience most of the jitter related issues are because people are using too small of audio buffer sizes that match the framing size of Speex - particularly in Windows. This isn't a problem with Speex, but as a programmer you should collect and append a few frames to match the size of your output audio frame buffer before attempting to play the sound. -----Original Message----- From:

This is a common problem. Sound cards run at different clock speeds (but your difference seems kinda high). What I'd suggest is dropping silence packets once in a while. .a -----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On Behalf Of Shantanu Gramsci Sent: Wednesday, December 21, 2005 2:05 PM To: speex-dev@xiph.org Subject: [Speex-dev] Bitrate

Hello everyone I've a following problem with the speex codec. I'm developing application that is supposed to read sounds from the microphone with DirectSoundCapture. Unfortunately when I want to compress and decompress the sound buffer from DirectSound I only have jitter. Maybe you have some code samples how to compress standard DS buffers: LPDIRECTSOUND8 lpDS = NULL;

Hi, A while ago I mentioned I was considering splitting libspeex into two parts. I'm attaching a patch that does exactly that, but I haven't applied it to svn because I first want to have some feedback. That patch leaves the codec in libspeex and moves the other stuff (jitter buffer, AEC, preprocessor, ...) to libspeexvoip (I'm open to suggestions on better names). Some of the things

Jim, Thank you very much for your suggestions. I managed to get the C55x code working on the simulator. I would like to port Speex both on a C5502 EVM and a C5509A EVM. As such, if you can provide me with the details of your port on the C5509A, it would be greatly appreciated. Furthermore, I am looking for some technical documentation on the CELP algorithms. I would like to better

Oh, I forgot to mention one thing. I currently append a few bits of custom information to each speex packet which I fetch out with things like speex_bits_unpack_unsigned(&sjJitter.current_packet, 1); It would be very usefull if the jitter buffer didn't actually decode the packet, but instead returned a pointer to it (or NULL if you should play silence and -1 if you should repeat the

Hi everyone, Judging from the lack of complaints about the split, I think I can go safely forward with that. The only thing to decide is the name of the new library that will contain the preprocessor, the AEC, jitter buffer, and resampler. So far, I've got a few potential names: - libspeaks - libspeexproc - libspeexvoip - libspeech - libspoke Let me know (in private so as to not flood the

Thank you for a very good explanation which shed light on some of the questions that I had after reading the source code. Reading your text however, I wonder if I'm perhaps missing an important point on the proper use of the jitter buffer: ... > Now, clearly, if early_ratio is high and late_ratio is very > low, the buffer is buffering more than it needs to; it will > skip a frame

Hi, We are planning to use Speex as the speech codec for a VoIP application. 1) May I know how Speex compared with GIPS codec? It seems that Google, Yahoo, and Skype are licensing from GIPS. Are there any good benchmarking or fair comparisons? 2) In particular, how is the jitter buffer control for Speex in response to intermitent poor connection hiccups? Is it robust enough to smooth out

Dear all, I need to use echo cancellation in a VOIP application using speex narrowband codec.I am facing some problems with echo cancellation. It will be very helpfull if someone can put some light on the following issues. 1: =================== -- When call is started EC works fine for first few seconds. Bur after a certain period [usually 20 sec or more] it does not work or works very bad

Hi, I found this message concerning VAD and was wondering whether VAD has been ported to fixed-point in the latest version? Thanks, SingHui ---------- Forwarded message ---------- From: Jean-Marc Valin <Jean-Marc.Valin@usherbrooke.ca> Date: Jul 22, 2005 1:02 AM Subject: Re: [Speex-dev] Fixed-point To: gue baja <gue_baja@yahoo.com> Cc: speex-dev@xiph.org Hi Baja, Here's a quick

> Hello, Hi :) First off, could you try to set your email client to break long lines before transmitting? In my (somewhat outdated) pine, the lines appear as VERY long lines when I try to reply, making it hard to read :) Minor detail though, I should probably fix pine. Some day. > The way you describe how the jitter buffer should be implemented makes me > wonder: How does the

Hi everyone, In the 1.1.x branch, I've kept adding more stuff to libspeex: preprocessor, AEC, etc. I'm now considering moving all those to a separate library (libvoip, libspeech, whatever). Anyone on this list has good reasons I should consider for either splitting or not splitting libspeex? Cheers, Jean-Marc

Hi Andras, others, Andras Kadinger <bandit <at> surfnonstop.com> writes: > > > Daniel, > I recommend you to start from a simple case and gradually progress > towards your goal. > Can you make things work with the "Speex in a Disco" (Example 6) > testcase at http://ns.surfnonstop.com/~bandit/speex/echocard1/ ? > These files

>I remember?I had to expose the echo cancelation level implementing a get_echo_level( ) function based on this: >http://lists.xiph.org/pipermail/speex-dev/2008-September/006889.html This is really a good idea to determine the frequency difference between capture and play of the sound card. But it need constant far-end voice and a long time because it must repeat the process of

Hi All, We are using speex for linphone in blackfin fixed point platform. The speex version we are using is svn-14525. If we don't disable vbr by "--disable-vbr", the sound quality will be very bad in linphone. But if we disable it, sound will be perfect. I noticed in the speex.org, there is a comment like this: > >Speex 1.2beta3 is out >December 11, 2007 >The most

> Hi, > > It could (and should) definitely be done, but it's somewhere > (unfortunately not on top) on my TODO list. I think the simplest way > (until I really integrate the preprocessor with the codec, which will > take a while) to do it is to add a speex_encoder_ctl() call as you > proposed. I'd call it SPEEX_SET_EXT_VAD_STATUS and I'm willing to merge > a

Hi, Just wanted to know, is speex still being developed? I was going through the manual and its dated December 2007 for speex version 1.2 beta 3. As far as I can see there doesnt seem to be any new versions since then, is there any development work on speex? regards, James -------------- next part -------------- An HTML attachment was scrubbed... URL: