similar to: How to set speex input paras

How speex cope with the difference between the codec frame length ,ie.160, and denoising algo frame length? ? The frame length of denoising algo? must be different from that of codec. Although I do not know the exact frame lenth is , it must be a certain power of 2,say 128 or 256.not 160. Then how speex handle it? ? Cheers HyeeWang

Hi Steve, Have you read this paper? (Heping Ding, David I. Havelock, Drift-Compensated Adaptive Filtering for Improving Speech Intelligibility in Cases with Asynchronous Inputs. EURASIP J. Adv. Sig. Proc. 2010:) Let me call is paper-Drift. It provided a method to evaluate Relative Sample Offset (RSO, d[i]) which is omitted in the microsoft paper (Challenges and Solutions for Designing Software

OK. Different mailing lists are set up differently. This list is unusual because your answers only go to the person who replied to you. So if you want the other people on the listserv to see your answer, you should make sure that Speex-dev at xiph.org<mailto:Speex-dev at xiph.org> is added to the TO: field of your outgoing message. Hopefully someone else will also attempt to answer your

Jean-Marc Valin wrote: > Hi, > > I just released unstable version 1.1.2 that contains more fixed-point > work. Though it's still not 100% complete, enough have been done to make > it run in real-time on ARM. In order to do that, compile with > --enable-fixed-point --enable-arm-asm. All narrowband modes work in > real-time with complexity 1 (some work with higher

Hello, I am using Cent OS 5.2 . i want to create one master image of whole running OS becoz i have to do same installation many times on different machines. so any one suggest me to how create and run the image of whole OS at the time of installation or , which escape me to install each and every thing many times. i would not like to create image using dd. any suggestions ?? - - Thanks and

I am swetha doing final btech.i want to do project regarding ssh.I want to port this openssh open source code to vxworks-6.4.Is it possible to port it to vxworks-6.4?why becoz there is no support for ssh in vxworks-6.4.And also i got key files /etc/ssh/ in linux.how can i got such type of files in vxworks-6.4.Please give me a suggestion regardind this. Thank you swetha.

Hello, I just signed up to this mailing-list (note: my first mailing list at all), because I'm having some problems related to speex. Let me just introduce you to what I'm doing. I am writing a short (really short) paper about VoIP techniques, especially audio codecs for speech. I pointed out basic technologies behind audio codecs; vector quantization, lpc, long-term prediction and some

Hello, Anyone else experiencing problems with the latest spandsp (pre3) and last libtiff beta? I'm getting 8 bytes long file, with the TIFF header only during such connection: -- Accepting call from 'XXXXXXX' to 'YYYYYY' on channel 0/2, span 1 -- Executing SetVar("Zap/2-1", "FAXFILE=/tmp/foch.tif") in new stack -- Executing

I have two pairs of time intervals: coarse- and fine-grained. They're components of their respective dataframes, looking like, coarse: endtime starttime 1 t1_end t1_start 2 t2_end t2_start ... fine: is the same, except that its intervals presumably fall into the coarse's enclosing ones. The problem is to partition

hi, Paul Novodon Maybe the same question happened to me some yrs ago when I was testing a PBFDAF AEC filter. Could you provide me the recorded file to me to study? Including: the working two files ( far end signal and far end echo + near end signal) the not working two files ( far end signal and far end echo + near end signal) HyeeWang > ------------------------------ >

Hi i have huge traffic going to internet at this moment iam using cisco to do QoS after reading lartc i would like to shift the setup to linux can you any one tell me i have 45mb internet make 40mb for all traffic, and 5mb for voice if Voice not used 45 MB for Data any sample configuration , or example will be good to start with becoz its very high traffic .. so suggest me what server

Hi, I have declared my sip users call-limit=2 and type=friend. When any user recieves a waiting call while already in a conversation, the peer call counter is set to 2.The problem is that, the counter is not reset to zero after hangup and becoz of this the user is not able to recieve any call anymore even if s/he has hungup. the asterisk cli displays the following error. [Mar 17 16:15:10]

Dear Steve after a lot of compile from spanDSP first version to K and after i have compile * HEAD and Stable i cant understand where im wrong... - Mandrake 9.2 - libtiff3-3.5.7-11mdk I have add Header from your FTP but the result is this error.. Can you give me your opinion? Thanks in advance Dimitri -------- -- Executing RxFAX("Zap/35-1", "/home/user/testfax.tif") in

On the /proc/zaptel it was lost interrupts, you haven't got any so that's good! On opencall.org there's a known issues link and that mentions some fax machines that have issues, might be worth a quick check there. I can receive from our fax machine this end, if you'd like I'll send you a test fax if you send me your fax number, if you can receive from us and tpc then it's

hi actually im trying to add feature to ices2 to able to stream mix file format such as ogg and mp3 in the same playlist.m3u without extra decoding and encoding that function like ezstream. the reason i dont want to use ezstream as source client becoz after going thru decoding and encoding, the audio quality drops. Therefore im trying to find out is there any easy way out to add plug-in to

I've been trying to set up incoming faxes using spandsp with a HFC card. Unfortunately, incoming faxes are of very poor quality, the pages are not transferred wholly (sometimes only a bit of a page is transferred etc.). From what I've read, I may have troubles with correct "timing" (and need to set up ztdummy etc.). On the other hand, it is impossible to set ztdummy if one

This is resolved: apt-get remove libspeex-dev cd ~/src/speex-1.2rc1/ ./configure --prefix=/usr make; make install cd ../ffmpeg ./configure --enable-libspeex make; make install worked; then I was able to decode a Speex .flv file: ~/flvs$ ffmpeg -i SpeexQ6R16Efalse.flv foo.wav FFmpeg version SVN-r17174, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --enable-libspeex

DEAR: I SEEM A BUG FOR speex-1.2rc1.tar.gz ON BUILD IN USING IT. PLEASE LOOK: ===================================================================== make all-recursive make[1]: Entering directory `/tmp/speex-1.2rc1' Making all in libspeex make[2]: Entering directory `/tmp/speex-1.2rc1/libspeex' /bin/sh ../libtool --mode=link gcc -g -O2 -fvisibility=hidden -O3 -msse -o testenc

Hi, I just sent this to Steve Underwood, but then found a bunch of posts on the mailing list about similar issues.. does anyone have the fix? I'm running asterisk CVS-HEAD-06/28/04-18:13:13, spandsp 0.0.1k, libtif 3.5.7 one thing i just noticed is that calls come in with format '72' which is G711A-law or LinearPCM.. it uses PCM for the call, i assume this is ok the results of RxFAX

Hello, We are working on the Radio station project and we are trying to implement the Live Video Streaming module using Icecast server with VLC player. We have installed and configured the Icecast into our server. Also, we have installed the VLC player in our local desktop computer. The USB webcam(200 CMOS UCAM - DLC220T) has been connected into my machine. Now we are trying to live video