similar to: need a help for using AEC

The AEC is not designed to cancel sinusoids. Try with speech. Jean-Marc Johnson Peng a ?crit : > speex-devDear Jean-Marc Valin I got some problems with evaluating the > AEC module of speex. I wrote a test main function and compiled it > with the speex lib in VC6.0, it initialized the AEC state and called > the AEC main function in the same way as what was done in testecho.c. >

Hi all, I have a query. I developed NLMS acoustic echo cancellation algorithm for real time voip application and using general purpose processor. Used more filter length (4096/2048) for NLMS and faced CPU consumption.With less filter length unable to estimate the exact echo signal and the quality of the speech also getting degraded. Could any one please suggest me the filter length

Hello Jean-Marc Valin and all the Speex-dev team, Today I have a little question about patents and the AEC. I' am not a lawyer, but after a deep research in esp@cenet.com, I found that in August 2003, an AEC called "Double talk insensitive NLMS algorithm" (DNLMS) have been patented by Gary Q Jin, Gordon J Reesor, and others (US 6.608.897 B1). It uses a continuous adaptation step

I would like to thank all the people who have contributed to this very fine project. Great work! I''ve encountered some strange results while examining the term frequency of one of my indexed documents. The indexed terms seem to vary for the very same document depending on the presence or absence of completely unrelated operations in the code, so the resulting term frequency changes, too.

Hello, I tried to use speex echo canceller for a live performance setup to cancell feedback. My program buffers input until collecting a speex frame size of samples, then calls speex_echo_cancellation passing this buffer as input signal and previous speex output as echo signal, then immediately plays current speex output. First thing I noticed was huge high pass filtering. Then I found out there

Hi All I am developing a client that uses libjingle to do xmpp stuff with ejabberd. I can also make audio calls between those clients. What I am trying to archive now is to send calls to pstn using jingle. I was told in the jingle-dev community that asterisk can do that. Is there any way to send jingle audio calls to asterisk and will it understand them ? If yes..can I forward those calls to PSTN

hi,friends I have downloaded the oggdropXPdV1.8.9-1.2.0 ,and opened the oggdropXPd.dsw in vc6.0. It prompt that some projects can't be found. One of them is static_libsamplerate.dsp which default path is .\libsamplerate-0.1.2\Win32\static_libsamplerate\static_libsamplerate.dsp. then I download the libsamplerate-0.1.2.tar.gz,and uncompressed ,there really have a win32 folder, but there

Hullo, all. I diligently consulted the suggested reading on the way in: > Welcome to the rsync@lists.samba.org mailing list! PLEASE READ THIS > IMPORTANT ETIQUETTE MESSAGE BEFORE POSTING: > http://www.tuxedo.org/~esr/faqs/smart-questions.html 'Twas a quick read, given that I'm the co-author. ;-> I recently received an odd inquiry from someone I didn't know at all,

Greetings! I now found someone to test gtalk with and found out, that app_jack has a problem here. My voice gets transmitted fine, but I only get white noise from the other party. I tried to set my JACK samplerate to 8000 to make sure it's no libresample problem, the results were the same. My setup is: Linux Debian Lenny Kernel: 2.6.30.4 PREEMPT (self-built) JACKd: jackd version

2014-05-03 18:40 GMT+02:00 Brad Isbell <brad at musatcha.com>: > The general idea was to make a really simple source client that could be > launched with a single click (such as from a hosting provider's admin > panel). The client accepts parameters on the query-string for configuring > so that in a real-world environment, the user would not have to configure > anything.

there is a DLL in bin directory of asp project. this DLL was written by VC6.0. I don't have source code of this DLL. The mono report a error "can't not find meta data in the DLL", I think that this DLL is not written by C#, so mono can't load it, I want to use wine, but I don't know how to do it, can anyone help me? Thansk!

I am trying to test drive ferret on a ubuntu dapper installation. I have a ferret-test.rb file like: ----begin--------- require ''ferret'' include Ferret index = Index::Index.new(:path => ''/opt/search-index'') ----end----------- When i issue: ruby ferret-test.rb, I get: ferret-test.rb:1:in `require'': no such file to load -- ferret (LoadError)

I'm having trouble getting the echo canceller to work and I'm hoping some one can get me started in diagnosing what's going on. The symptoms I'm seeing are these: 1. If the NES is silence (i.e., no echo, no speech), the ec puts the FES back into the audio. 2. If the NES is similar to the FES (same speech words), the ec doesn't seem to remove the echo. 3. For case #2, I pushed

Hi, I've got an AEC Star T4110 connected with an RS232 port by a megatec cable. On the list of supported devices (http://www.networkupstools.org/stable-hcl.html) there is only one AEC device, so I tried genericups with upstype=21. The problem is, that the UPS (while being on-line) reports constantly being on battery. # upsc aec ups.status OB My question is: What is the recommended procedure

Hi! In a message posted on May 31 Jean-Marc wrote in a topic about AEC: > - set sampling rate to 8 kHz (at least for now) Does it mean that echo cancellation algorithm won't work in wideband mode? I can't get aec to work well in my application and I am wondering whether it's not a matter of using wideband instead of (preferred?) narrowband mode? Thanks in advance for any

Hi I was trying out the AEC in speex. It would be helpful if some of you can answer these querries 1.After the error values are computated in teh daptation process a check -if filter is completely mis-adapted - is performed. How could you explain the criteria used for this check. 2.what exactly is meant by over-cancellation and under-cancellation ? 3.Also it was seen that without the

I've recently made changes to the AEC. Please try the code in SVN and see if it works better. Jean-Marc Le jeudi 03 novembre 2005 ? 22:36 -0800, Jason Harper a ?crit : > I've tried some further debugging to see what mdf is > actually doing. Instead of sending: > tmp_out = (float)ref[i] - st->y[i+st->frame_size] > to the output, I just sent >

> To everyone on the list: do *NOT* attempt to do echo cancellation with > signals sampled using different clocks. This will *NOT* work. Just a > 0.1% difference between the two sampling rates (it's sometimes worse > than that) means that the impulse response drifts by 8 samples every > second. There's just no way to efficiently track this. Or at least no > way that

OK, I'm tired of arguing. All those who think they're going to do AEC on different un-synced cards, just go for it. However, please do not complain and/or ask questions about that on this list. AEC is already a hard enough problem when you have a sane sound setup that I see no point in trying to do anything with drifting clocks. As for soundcards with different clocks for in and out, I

Hi: I have been trying to use speex for AEC. I have recorded two voice samples separately using audacity using the same soundcard. I reduced the gain of one of the samples and mix this signal with the other voice sample (saved as a raw sample, with frame size 64 and sampling rate of 8 KHz. I use the mixed one as a ref signal and the other unaltered signal as the echo. Am trying to run the