search for: vbr_qualiti

Hi, I'm having a bit of trouble with the ABR. I'm a developer of Mumble, a voicechat application for gamers, and up to now we've just used VBR quality to determine the bitrate. In an effort to make the resource requirements a little more determinable (and limitable) for hosting, I tried switching to ABR, as there doesn't seem to be a way to determine the absolute peak

Hi, I'm trying to cover all Speex configuration options to set quality/mode/bitrate in my application and I was wondering if the following all makes sense: - abr_bitrate - use average bitrate and set it using bitrate parameter - SPEEX_SET_ABR - cbr_bitrate - use constant bitrate and set it using bitrate parameter - SPEEX_SET_BITRATE - cbr_mode - use constant bitrate and

Hi guys I have tried compiling this attached code, I made all the buffers 320, there is no trace of a 160 buffer, but I get a " SpeexEncoder requires 320 samples to process a Frame, not 160" error. Maybe there's something I'm missing, here's my code: import java.io.IOException; import java.io.FileOutputStream; import java.io.File; import

Hi, I am looking how to setup speex codec in codecs.conf for 8 Kbps and 6 Kbps. In config file are many parameters for setting. I don't know what is need to change for narrowbad like 8 Kbps and 6 kbps. Any suggestion? [speex] ;0-10 quality => 4 ;0-10 complexity => 4 ; true / false enhancement => true ; true / false vad => false ; true / false vbr => false ; 0 = off, otherwise,

Jean-Marc Valin wrote: >> gcc version 3.4.5 (mingw special) > > Can you try 4.0 or 4.1 just to be sure? mingw (native gcc for win32) doesn't officially support using gcc 4.0 and 4.1 yet (apparantly there have been some issues), so there are no binary packages. But if you think it helps, I can compile gcc 4.1 and give it a shot. >> Compile flags: >> DEFINES +=

Great, thanks! That patch solved it. VBR without DTX seems to work perfectly now. If I enable DTX, there's still a little bit of buzz but it doesn't seem to last very long when it happens. I appreciate the quick fix! Tom Jean-Marc Valin (jean-marc.valin@hermes.usherb.ca) wrote: > > Yep, you found a bug and here's the patch. It should solve your problem > (using VBR

On Mon, 2005-09-05 at 10:29 +1000, Jean-Marc Valin wrote: > > attached is a patch fixing some build warnings with ip3k-elf-gcc > > (gcc 3.4-20030722-Ubicom-63-1211). not all fixes are elegant, > > please review carefully before applying.. > > Euh, could you explain what's wrong with "start" and "end"? Also, AFAIK, > (int)floor(something) is

Hi, I'm updating speex support in sweep to RC2 and I'd like some advice on encoder options, and would appreciate some testing of the user interface. a screenshot of the updated speex encoding dialog: http://www.vergenet.net/~conrad/tmp/sw_speexRC2_1.png you can grab a tarball of sweep for testing here: http://www.vergenet.net/~conrad/tmp/sweep-0.8.0-speexRC2.tar.gz general

Hi Jean-Marc and All, Two things need your confirmation/suggestion. 1) First, I think there is a bug in libspeex/lsp.c line 512. /* hard limit ak's to +/- 32767 */ if (a < -32767) a =32767; // This line should be changed to if (a < -32767) a = -32767; if (a > 32767) a = 32767; ak[j-1] = (short)a; 2) About the RTP wrapper for VoIP I'd like to use the

In a word, I don't what's the standard of speex payload format. The file doc/rtp.txt is for what? Is it not for rtp payload? I find that rtp.txt is more detail that draft02.txt Which rtp docment should be followed? Anyone else has written the RTP wrapper already? Lianghu On 11/22/06, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > > > 1) First, I think there is

if the new draft in the manual is used. I don't find how to tell the decoder which mode(NB/WB/UWB) is used in the encoder. The RTP header don't contain the mode field and I don't find the mode information in the coded frame either. Does this mean we have to use NB decoder in all cases? Lianghu On 11/22/06, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > >

Hi Jean-Marc, Hi Jim, I have also seen some problems with the 1.1.8 release on the C55x. So far I have boiled down the issues to the following: 1) We need our own "fixed_xx.h" header file. I don't know why, and haven't had time to investigate, but there is a definite improvement when I use the attached fixed_c55x.h file which has turned all the maths into inline functions.

Jean-Marc Valin (jean-marc.valin@hermes.usherb.ca) wrote: > > > I've been using Speex in my voice-over-IP program on Win32, in > > wideband (16kHz) mode. I just starting using VBR recently and > > have run into something that might be a problem within Speex: > > Are you turning on DTX in addition to VBR? Also, what version are you > using. As of 1.0, DTX is

Hi Conrad, Thanks a lot for the info. This is exactly what I needed. We want to use speex in our IP telephony client and I was indeed trying to put together a GUI configuration screen. Best regards, Chris > On Tue, May 20, 2003 at 06:04:39PM +0200, Chris Flerackers wrote: >> Hi, >> >> Is there are document where the interaction between all the configuration >> options

hi attached is a patch fixing some build warnings with ip3k-elf-gcc (gcc 3.4-20030722-Ubicom-63-1211). not all fixes are elegant, please review carefully before applying.. /alfred -------------- next part -------------- A non-text attachment was scrubbed... Name: warnings.diff Type: text/x-patch Size: 17666 bytes Desc: not available Url :

> there is nothing wrong with "start" and "end", but in the Ubicom OS > there is a global function called "start()" which gives the warning: > > libspeex/ltp.c:573: warning: declaration of "start" shadows a global declaration > build/include/UbicomSDK.h:5651: warning: shadowed declaration is here Since it's just a warning and just on

Ok, I just about have this working - now I am able to get both my liveice stream to connect properly to icecast, but I no longer get a "no encoder" error on connecting to icecast w/winamp! However, nothing plays and I get a timeout pretty soon. How can I check and make sure that liveice is streaming my static files? here's the cfg - do I have the correct settings for streaming a

Dear All, If there is anyone using liveice here....some questions for... 1.What is the latest version of liveice? Is it the one on sourceforge or the tar.gz on your website? 2. i have icecast running nicely and am able to do simple playlists / grab audio from a mike via ices0.4 and darkice. I am now experimenting with liveice. 3. What is the access url for others to listen to my stream. Where is

> 1) First, I think there is a bug in libspeex/lsp.c line 512. > > /* hard limit ak's to +/- 32767 */ > if (a < -32767) a =32767; // This line should be changed to if > (a < -32767) a = -32767; > if (a > 32767) a = 32767; > ak[j-1] = (short)a; Oops. Thanks for pointing that out. It's fixed in svn. > 2) About the RTP wrapper for VoIP

lianghu xu wrote: > In a word, I don't what's the standard of speex payload format. > The file doc/rtp.txt is for what? Is it not for rtp payload? > I find that rtp.txt is more detail that draft02.txt > > Which rtp docment should be followed? > Anyone else has written the RTP wrapper already? Oh, I see. doc/rtp.txt was a very, very early draft. See the manual for a